ADMINISTRATION GUIDE Cisco Small Business Pro SPA2102, SPA3102, SPA8000, PAP2T, WRP400 Analog Telephone Adapters
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Contents About This Document ix Chapter 1: Introducing Cisco Small Business Analog Telephone Adapters 16 Comparison of ATA Devices 17 ATA Connectivity Requirements 20 PAP2T Connectivity 21 SPA2102 Connectivity 22 SPA3102 Connectivity 23 SPA8000 Connectivity 24 ATA Software Features 25 Voice Supported Codecs 25 SIP Proxy Redundancy 27 Other ATA Software Features 27 Chapter 2: Basic Administration and Configuration 35 Basic Services and Equipment Required 35 Downloading Firmware
Contents Reboot URL Provisioning Your ATA Device 44 44 Provisioning Capabilities 44 Configuration Profile 45 Chapter 3: Configuring Your System for ITSP Interoperability Network Address Translation (NAT) and Voice over IP (VoIP) 47 47 NAT Mapping with Session Border Controller 48 NAT Mapping with SIP-ALG Router 48 Configuring NAT Mapping with a Static IP Address 48 Configuring NAT Mapping with STUN 50 Determining Whether the Router Uses Symmetric or Asymmetric NAT 52 Firewalls and SIP 5
Contents Using a Mini-Certificate 74 Generating a Mini Certificate 75 SIP Trunking and Hunt Groups on the SPA8000 About SIP Trunking 78 Setting the Trunk Group Call Capacity 80 Inbound Call Routing for a Trunk Group 80 Contact List for a Trunk Group 81 Outgoing Call Routing for a Trunk Group 83 Configuring a Trunk Group 84 Trunk Group Management 85 Setting the Hunt Policy 86 Additional Notes About Trunk Groups 87 Chapter 5: Configuring Music on Hold Using the Internal Music Source for
Contents Configuring VoIP Failover to PSTN 102 Sharing One VoIP Account Between the FXS and PSTN Lines 103 Other Options 104 PSTN Call to Ring Line 1 104 Symmetric RTP 104 Call Progress Tones 105 Call Scenarios PSTN to VoIP Call with and Without Ring-Thru 106 VoIP to PSTN Call With and Without Authentication 106 Call Forwarding to PSTN Gateway 109 Appendix A: ATA Routing Field Reference Router Status page 111 111 Product Information section 112 System Status section 112 WAN Setup pa
Contents Port Forwarding Settings section 119 DMZ Settings section 119 Miscellaneous Settings section 120 System Reserved Ports Range section 120 Appendix B: ATA Voice Field Reference Info page 122 Product Information section 122 System Status section 123 Line Status section 123 System Information section (PAP2T) 126 PSTN Line Status section (SPA3102) 126 Trunk Status section (SPA8000) 129 System page 130 System Configuration section 130 Internet Connection Type section (PAP2T) 1
Contents Distinctive Ring/CWT Pattern Names section 150 Ring and Call Waiting Tone Spec section 151 Control Timer Values (sec) section 151 Vertical Service Activation Codes section 153 Vertical Service Announcement Codes section (SPA2102, SPA8000) 159 Outbound Call Codec Selection Codes section 159 Miscellaneous section 161 Line page Line Enable section 166 Streaming Audio Server (SAS) section 166 NAT Settings section 167 Network Settings section 168 SIP Settings section 169 Call Feat
Contents PSTN Line page (SPA3102) 190 Line Enable section 191 NAT Settings section 191 Network Settings section 192 SIP Settings section 193 Proxy and Registration section 195 Subscriber Information section 197 Audio Configuration section 198 Dial Plans section 201 VoIP-To-PSTN Gateway Setup section 202 VoIP Users and Passwords (HTTP Authentication) section 204 Ring Settings section 205 FXO (PSTN) Timer Values (sec) section 205 PSTN Disconnect Detection section 207 International
Contents Appendix C: Provisioning Reference (WRP400) 221 Appendix D: Troubleshooting 235 Appendix E: Environmental Specifications 239 PAP2T 239 SPA2102 240 SPA3102 240 SPA8000 241 WRP400 242 WRTP54G 242 Appendix F: Where to Go From Here 244 Product Resources 244 Related Documentation 245 Appendix G: Additional Information 247 Appendix H: Support Contacts 248 ATA Administration Guide viii
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.
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.
Preface Organization The information in this guide is organized into the following chapters and appendices: ATA Administration Guide Chapter Contents Chapter 1, “Introducing Cisco Small Business Analog Telephone Adapters” This chapter introduces the functionality of the ATA devices and describes the features that are available.
Preface Chapter Contents Appendix D, “Troubleshooting” This appendix provides solutions to problems that may occur during the installation and operation of the ATA devices. Appendix F, “Where to Go From Here” These appendices provide information about other resources that may be useful to you.
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. • Search multiple PDF files at once (for example, all PDFs in a specific folder or disk drive). • Perform advanced searches. Finding Text in a PDF Follow this procedure to find text in a PDF file.
Preface Finding Text in Multiple PDF Files The 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. STEP 1 Start Acrobat Professional or Adobe Reader. STEP 2 Choose Edit > Search, or click the arrow next to the Find box and then choose Open Full Acrobat Search. STEP 3 In the Search window, complete the following steps: a. Enter the text that you want to find. b. Choose All PDF Documents in.
Preface STEP 4 When the Results appear, click + to open a folder, and then click any link to open the file where the search terms appear. For more information about the Find and Search functions, see the Adobe Acrobat online help.
1 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-toend 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.
1 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).
Introducing Cisco Small Business Analog Telephone Adapters Comparison of ATA Devices 1 NOTE The information contained in this guide is not a warranty from Cisco. Customers 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. By implementing ATA devices with the SIP protocol, intelligent endpoints at the edges of a network perform the bulk of the call processing.
1 Introducing Cisco Small Business Analog Telephone Adapters Comparison of ATA Devices Figure 1 How ATAs Provide Voice Connectivity Ethernet/Wireless LAN WRP400, WRTP54G, and SPA2102 Fax (up to 4 SPA8000) DSL/cable modem Broadband router Internet Analog phone (up to 8 with SPA8000) Broadband router SPA8000, PAP2T SPA3102 PSTN ATA Administration Guide 187255-revised Ethernet/Wired LAN • The SPA3102 and SPA8000 act as SIP-PSTN gateways.
Introducing Cisco Small Business Analog Telephone Adapters ATA Connectivity Requirements 1 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.
1 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 2). Administrative IVR (Line 1 or Line 2) Line 1 Ethernet port IP Router (with hairpinning) or Broadband modem ISP Internet Line 2 LAN WAN PAP2T 187420 IP ITSP NOTE ATA Administration Guide • The IVR functions are accessed by connecting an analog telephone to Line 1.
1 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 Line 2) Line 1 Ethernet port IP Router (with hairpinning) or Broadband modem ISP Internet Line 2 LAN WAN IP LAN port Administration PC 187257 SPA2102 ITSP By default, the device attached to the LAN port is assigned the network address 192.168.0.
1 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 Line 2) Line 1 PSTN Ethernet port IP Router (with hairpinning) or Broadband modem ISP PSTN Line 1 LAN Internet WAN IP LAN port Administration PC 187259 SPA3102 ITSP By default, the device on the LAN port is assigned the network address 192.168.0.0 with a subnet mask of 255.
1 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).
Introducing Cisco Small Business Analog Telephone Adapters ATA Software Features 1 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. If necessary, explicit port ranges can be specified for SIP and RTP.
Introducing Cisco Small Business Analog Telephone Adapters ATA Software Features 1 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. See “ATA Voice Field Reference,” on page121. See also “Supported Codecs,” on page 54 for a list of which codecs are supported on each ATA device. Codec (Voice Compression Algorithm) Description G.
Introducing Cisco Small Business Analog Telephone Adapters ATA Software Features 1 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.
Introducing Cisco Small Business Analog Telephone Adapters ATA Software Features 1 Feature Description Modem and Fax Pass-Through • Modem pass-through mode can be triggered only by predialing the number set in the Modem Line Toggle Code. (Set in the Regional tab.) • FAX pass-through mode is triggered by a CED/CNG tone or an NSE event. • Echo canceller is automatically disabled for Modem passthrough mode.
Introducing Cisco Small Business Analog Telephone Adapters ATA Software Features ATA Administration Guide 1 Feature Description Adjustable Audio Frames Per Packet This feature allows the user to set the number of audio frames contained in one RTP packet. Packets can be adjusted to contain from 1–10 audio frames. Increasing the number of packets decreases the bandwidth utilized, but it also increases delay and may affect voice quality.
Introducing Cisco Small Business Analog Telephone Adapters ATA Software Features 1 Feature Description Signaling Hook Flash Event The ATA device can signal hook flash events to the remote party on a connected call. This feature can be used to provide advanced mid-call services with third-party-callcontrol.
Introducing Cisco Small Business Analog Telephone Adapters ATA Software Features ATA Administration Guide 1 Feature Description Calling Party Control Calling Party Control (CPC) signals to the called party equipment that the calling party has hung up during a connected call by removing the voltage between the tip and ring momentarily. This feature is useful for autoanswer equipment, which then knows when to disengage. CPC is configured in the Regional, Line, and PSTN Line tabs.
Introducing Cisco Small Business Analog Telephone Adapters ATA Software Features ATA Administration Guide 1 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. SIP over TLS relies on the widelydeployed and standardized TLS protocol.
Introducing Cisco Small Business Analog Telephone Adapters ATA Software Features 1 Feature Description Register Retry Enhancements The Register Retry Enhancements feature for SPA2102, SPA3102, and PAP2T devices adds flexibility to the delay timers that are activated when the SIP REGISTER of a device fails. Once a SIP REGISTER failure response code is sent, a delay timer is selected depending on the type of registration failure response code.
Introducing Cisco Small Business Analog Telephone Adapters ATA Software Features 1 Feature Description DHCP Renewal on Timeout SPA2102, SPA3102, and PAP2T voice devices typically 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.
2 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.
Basic Administration and Configuration Downloading Firmware 2 • 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. 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 Administration and Configuration Setting up Your ATA Device STEP 3 2 Use the administration computer to install the latest firmware: a. Extract the Zip file, and then run the executable file to upgrade the firmware. b. When the Firmware Upgrade Warning window appears, click Continue. c. In the next window that appears, enter the IP address of the ATA device, and then click OK. d. In the Confirm Upgrade window, verify that the correct device information and product number appear. Then click Upgrade.
Basic Administration and Configuration Using the Administration Web Server 2 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.
Basic Administration and Configuration Using the Administration Web Server 2 Connecting to the Administration Web Server To access the ATA administration web server, perform the following steps. STEP 1 Start Internet Explorer on a computer that is connected to the same network as the ATA device. STEP 2 Determine the address of the ATA device. a. Connect an analog telephone to the Phone 1 port of the ATA device. b. Press **** on the keypad to access the IVR menu. c.
Basic Administration and Configuration Using the Administration Web Server STEP 3 2 Complete the WAN configuration for DHCP, static IP addressing, or PPPoE. For DHCP: a. Select DHCP from the Connection Type 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. If your service uses a specific PC MAC address, then select yes from the Enable MAC Clone Service setting. d.
Basic Administration and Configuration Using the Administration Web Server 2 Registering to the Service Provider To use VoIP phone service, you must configure your ATA device to the Service Provider. STEP 1 Start Internet Explorer, connect to the administration web server, and choose Admin access with Advanced settings. STEP 2 Click Voice tab > Line N, where N is the line number that you want to configure. STEP 3 Enter the account information for your ITSP.
Basic Administration and Configuration Upgrading, Rebooting, and Resyncing Your ATA Device 2 NOTE If the device has more than one Line tab, each line tab must be configured 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.
Basic Administration and Configuration Upgrading, Rebooting, and Resyncing Your ATA Device 2 NOTE If the value of the Upgrade Enable parameter in the Provisioning page is No, you 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. If no protocol is specified, TFTP is assumed.
Basic Administration and Configuration Provisioning Your ATA Device 2 The profile-path is the path to the new profile with which to resync, for example: http://192.168.2.217admin/resync?tftp://192.168.2.251/spaconf.cfg Reboot URL The Reboot URL lets you reboot the ATA device. The Reboot URL is as follows: http://spa-ip-addr/admin/reboot NOTE The ATA device reboots only when it is idle. Provisioning Your ATA Device This section describes the provisioning functionality of the ATA device.
Basic Administration and Configuration Provisioning Your ATA Device 2 The ATA device supports up to 256-bit symmetric key encryption of profiles. For the initial transfer of the profile encryption key (initial provisioning stage), the ATA device can receive a profile from an encrypted channel (HTTPS), or it can resync to a binary profile generated by the Cisco-supplied profile compiler.
Basic Administration and Configuration Provisioning Your ATA Device 2 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_. For example, Line 1 Proxy is named Proxy_1_ in XML profiles.
3 Configuring Your System for ITSP Interoperability This chapter provides configuration details to help you to ensure that your infrastructure properly supports voice services.
Configuring Your System for ITSP Interoperability Network Address Translation (NAT) and Voice over IP (VoIP) 3 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.
Configuring Your System for ITSP Interoperability Network Address Translation (NAT) and Voice over IP (VoIP) STEP 3 3 Scroll down to the NAT Support Parameters section, and then enter the following settings to support static mapping to your public IP address: • Handle VIA received, Insert VIA received, Substitute VIA Addr: yes • Handle VIA rport, Insert VIA rport, Send Resp To Src Port: yes • EXT IP: Enter the public IP address for your router.
Configuring Your System for ITSP Interoperability Network Address Translation (NAT) and Voice over IP (VoIP) 3 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. This option is considered a practice of last resort and should be used only if the other methods are unavailable.
Configuring Your System for ITSP Interoperability Network Address Translation (NAT) and Voice over IP (VoIP) 3 Voice tab > SIP > NAT Support Parameters STEP 4 Click Voice tab > Line N, where N is the number of the line interface. STEP 5 Scroll down to the NAT Settings section. • NAT Mapping Enable: Choose yes. • NAT Keep Alive Enable: Choose yes (optional).
Configuring Your System for ITSP Interoperability Network Address Translation (NAT) and Voice over IP (VoIP) 3 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.
Configuring Your System for ITSP Interoperability Firewalls and SIP 3 STEP 6 Click Submit All Changes. STEP 7 View the syslog messages to determine whether your network uses symmetric NAT. Look for a warning header in the REGISTER messages, such as Warning: 399 spa "Full Cone NAT Detected.” Firewalls and SIP To enable SIP requests and responses to be exchanged with the SIP proxy at the ITSP, you must ensure that your firewall allows both SIP and RTP unimpeded access to the Internet.
4 Configuring Voice Services This chapter describes how to configure your ATA device to meet the customer’s requirements for voice services.
Configuring Voice Services Using a FAX Machine (SPA2102, SPA3102 or SPA8000) • G.726-40 • G.729a • G.723 4 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 • G.729a Using a FAX Machine (SPA2102, SPA3102 or SPA8000) Follow this procedure to optimize fax completion rates. NOTE T.38 Fax is only supported on the SPA2102, SPA3102, and the SPA8000.
Configuring Voice Services Using a FAX Machine (SPA2102, SPA3102 or SPA8000) STEP 3 STEP 4 4 To optimize G.711 fallback fax completion rates, set the following on the Line tab of your ATA device: • Network Jitter Buffer: very high • Jitter buffer adjustment: disable • Call Waiting: no • 3 Way Calling: no • Echo Canceller: no • Silence suppression: no • Preferred Codec: G.711 • Use pref. codec only: yes If you are using a Cisco media gateway for PSTN termination, disable T.
Configuring Voice Services Using a FAX Machine (SPA2102, SPA3102 or SPA8000) 4 Fax Troubleshooting If have problems sending or receiving faxes, complete the following steps: STEP 1 Verify that your fax machine is set to a speed between 7200 and 14400. STEP 2 Send a test fax in a controlled environment between two ATAs. STEP 3 Determine the success rate.
4 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 Tab Description and Value Caller ID Method Regional 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.
4 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) – (c). All CID methods can be applied for this type of CID. • On Hook Caller ID Not Associated with Ringing — This feature is used to send VMWI signal to the phone to turn the message waiting light on and off (see Figure 1 (d) and (e)).
Configuring Voice Services Silence Suppression and Comfort Noise Generation 4 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. VAD uses a sophisticated algorithm to distinguish between speech and non-speech signals. Based on the current and past statistics, the VAD algorithm decides whether or not speech is present.
Configuring Voice Services Configuring Dial Plans 4 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. This section includes information that you need to understand dial plans, as well as procedures for configuring your own dial plans.
4 Configuring Voice Services Configuring Dial Plans Digit Sequence Function 0 1 2 3 4 5 6 7 8 9 0 * # Enter any of these characters to represent a key that the user must press on the phone keypad. x Enter x to represent any character on the phone keypad. [sequence] Enter characters within square brackets to create a list of accepted key presses. The user can press any one of the keys in the list. .
4 Configuring Voice Services Configuring Dial Plans Digit Sequence Function , Enter a comma between digits to play an “outside line” dial tone after a user-entered sequence. (comma) EXAMPLE: 9, 1xxxxxxxxxx An “outside line” dial tone is sounded after the user presses 9, and the tone continues until the user presses 1. ! (exclamation point) Enter an exclamation point to prohibit a dial sequence pattern. EXAMPLE: 1900xxxxxxx! The system rejects any 11-digit sequence that begins with 1900.
Configuring Voice Services Configuring Dial Plans • 4 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) 9, xxxxxxx After a user presses 9, an external dial tone sounds. The user can enter any seven-digit number, as in a local call.
Configuring Voice Services Configuring Dial Plans • 4 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 ) 9, 1 900 xxxxxxx ! This digit sequence is useful if you want to prevent users from dialing numbers that are associated with high tolls or inappropriate content, such as 1-900 numbers in the U.S.. After the user press 9, an external dial tone sounds.
4 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.
Configuring Voice Services Configuring Dial Plans 4 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. Unless you have a special dial plan string to allow a null entry, the call is rejected. The default value is 5.
Configuring Voice Services Configuring Dial Plans 4 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. Unless the user enters another digit within the specified number of seconds, the entry is evaluated as incomplete, and the call is rejected. The default value is 10 seconds.
Configuring Voice Services Configuring Dial Plans 4 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. s: The number of seconds; if no number is entered after S, the default timer of 5 seconds applies. Examples for the Interdigit Short Timer • Set the timer for the entire dial plan.
Configuring Voice Services Configuring Dial Plans 4 Editing Dial Plans You can edit dial plans and can modify the control timers. STEP 1 Start Internet Explorer, and then enter the IP address of the SPA9000. Click Admin 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. STEP 1 Connect to the administration web server, and choose Admin access with Advanced settings.
Configuring Voice Services Configuring Dial Plans STEP 4 ATA Administration Guide 4 Enter the desired values in the Interdigit Long Timer field and the Interdigit Short Timer field. Refer to the definitions at the beginning of this section.
Configuring Voice Services Secure Call Implementation 4 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 • ”Using a Mini-Certificate” section on page 74 • ”Generating a Mini Certificate” section on page 75 NOTE This is an advanced topic meant for experience installers. See also the LVS Provisioning Guide.
Configuring Voice Services Secure Call Implementation 4 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. The Mini-Certificate is valid if it has not expired, and it has a valid signature.
Configuring Voice Services Secure Call Implementation STEP 2 4 The caller sends the “Caller Final” message to the called party with the following information: • Message ID (4B) • Encrypted Master Key (16B or 128b) • Encrypted Master Salt (16B or 128b) 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.
4 Configuring Voice Services 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. Use the following URL: http://www.cisco.com/web/partners/sell/smb/products/ voice_and_conferencing.html#~vc_technical_resources NOTE The partner sites require a logon.
Configuring Voice Services Secure Call Implementation 4 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: Sm9lIFNtaXRoAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAxNDA4NTU1MTIzNAAAAAAAMDAwMDAwMDEw MTM00OvJakde2vVMF3Rw4pPXL7lAgIagMpbLSAG2+++YlSqt198Cp9rP/ xMGFfoPmDKGx6JFtkQ5sxLcuwgxpxpxkeXvpZKlYlpsb28L4Rhg5qZA+Gqj1hDFCmG6dffZ9SJhx ES767G0JIS+N8lQBLr0AuemotknSjjjOy8c+1lTCd2t44Mh0vmwNg4fDck2YdmTMBR516xJt4/ uQ/ LJQ
Configuring Voice Services SIP Trunking and Hunt Groups on the SPA8000 4 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. You can configure up to four trunk groups for the purpose of inbound call routing and outbound caller identification.
4 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. Fax PBX System Fax SPA8000 Integrated Access Device Internet ITSP PBX System The SPA8000 offers four trunk groups, numbered T1, T2, T3, and T4.
4 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.
Configuring Voice Services SIP Trunking and Hunt Groups on the SPA8000 4 NOTE Although the figure shows only one ITSP account, each standalone line and each 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. You can configure a trunk group’s call capacity parameter to meet the requirements of the ITSP.
Configuring Voice Services SIP Trunking and Hunt Groups on the SPA8000 STEP 5 4 If the call is picked up by the PBX, the Line UA replies 200 OK with SDP to the internal Proxy. The Trunk UA in turn replies 200 OK to the ITSP and relay the Line SDP in the 200 OK message also. If all goes well, the Line UA and the ITSP equipment start exchanging RTP packets afterwards. Contact List for a Trunk Group The hunting process for incoming calls is controlled by the Contact List.
Configuring Voice Services SIP Trunking and Hunt Groups on the SPA8000 4 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. If interval is *, the hunt is stopped at the first line that starts ringing, and rings the line until it answers, or the caller hangs up, or the line's ringer times out.
Configuring Voice Services SIP Trunking and Hunt Groups on the SPA8000 4 • ?,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).
4 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. Before you begin this procedure, determine which lines you want to associate with each trunk group that you are configuring.
4 Configuring Voice Services SIP Trunking and Hunt Groups on the SPA8000 STEP 3 Enter the settings for each trunk group, as needed: a. Click Voice tab > Tn, where n represents the trunk group number (T1 ... T4). b. Enter the account information in the Subscriber Information section. • Display Name: The Caller ID that you want to use for outbound calls on this line • User ID: Your account number with the ITSP (usually the telephone number) • Password: Your password for this ITSP account c.
Configuring Voice Services SIP Trunking and Hunt Groups on the SPA8000 4 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.
Configuring Voice Services SIP Trunking and Hunt Groups on the SPA8000 4 Additional Notes About Trunk Groups This section includes information about other topics that may be of interest when you are configuring trunk groups: ATA Administration Guide • 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.
5 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 • “Configuring a Streaming Audio Server,” on page 90 Using the Internal Music Source for Music On Hold An internal music source with the user ID imusic is available. It plays an internally stored music file repeatedly.
Configuring Music on Hold Using the Internal Music Source for Music On Hold 5 STEP 2 Start Internet Explorer, and then enter the IP address of the telephone. The Telephone Configuration page appears in a separate browser window. STEP 3 Click Admin Login, and then click Advanced. STEP 4 Click the Ext 1 tab. STEP 5 Scroll down to the Call Feature Settings section. STEP 6 Enter the following value in the MOH Server field: imusic STEP 7 Click Submit All Changes.
Configuring Music on Hold Configuring a Streaming Audio Server STEP 6 5 • path: The location and name of a music file in the correct format • For example, if the computer local IP address is 192.168.0.5, the directory is named musicdir, and the converted music file is named jazzmusic.dat , then you would enter the following URL: tftp://192.168.0.5:69/musicdir/ jazzmusic.dat Click Submit All Changes. The unit reboots. Then the unit downloads the file and stores it in flash memory.
5 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. NOTE Each SAS server can maintain up to five simultaneous calls. If the second line on the unit is disabled, then the SAS line can maintain up to 10 simultaneous calls.
Configuring Music on Hold Configuring a Streaming Audio Server 5 Configuring the Streaming Audio Server Use the following procedure to configure an SAS with an external music source. STEP 1 Connect an RJ-11 adapter between the music source (a CD player or iPod, for example) and an FXS port. STEP 2 Start Internet Explorer, connect to the administration web server, and choose Admin access with Advanced settings. STEP 3 Configure the FXS port: a.
Configuring Music on Hold Configuring a Streaming Audio Server 5 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: STEP 1 Power off the ATA device. STEP 2 Connect a phone to the port and make sure the phone is on-hook. STEP 3 Power on the ATA device.
6 Configuring the PSTN (FXO) Gateway on the SPA3102 This chapter describes how to configure the PSTN gateway on the SPA3102.
Configuring the PSTN (FXO) Gateway on the SPA3102 How VoIP-To-PSTN Calls Work 6 • 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. A second VoIP account can be configured to support PSTN gateway calls exclusively.
6 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. Otherwise, it compares the with the User ID parameter of the PSTN Line. If they are the same, the ATA device interprets this as a request for two-stage dialing (see the ”Two-Stage Dialing” section on page 97).
Configuring the PSTN (FXO) Gateway on the SPA3102 How VoIP-To-PSTN Calls Work 6 NOTE If Authentication is disabled, a default dial plan is used for all unknown VoIP users. 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. To invoke two-stage dialing, the VoIP caller should INVITE the PSTN Line without the user-id in the Request-URI or with a user-id that matches exactly the of the PSTN Line.
6 Configuring the PSTN (FXO) Gateway on the SPA3102 How PSTN-To-VoIP Calls Work Table 2 Parameters for Two-Stage Dialing Parameter Web Page Description Values VoIP Caller 1/2/ 3/4/5/6/7/8 PIN PSTN Line The PIN for VoIP Caller 1, 2, 3, 4, 5, 6, 7, or 8. 31-character string VoIP Caller 1/2/ 3/4/5/6/7/8 DP PSTN Line Specifies which dial plan to be used for this VoIP caller. If 0, dial plan processing is disabled; the given target number is dialed to the PSTN as is.
6 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.
Configuring the PSTN (FXO) Gateway on the SPA3102 How PSTN-To-VoIP Calls Work 6 Parameter Web Page Description Values Disconnect Tone: PSTN Line Tone Script of the disconnect tone to detect. The ATA device supports two frequency components. If the tone has only one frequency, use the same value for both frequencies. ToneScript Each cadence segment must have the same frequency. The default is 480@30,620@30;4(.25/ .25/1+2)” The level value is the threshold to detect each tone.
6 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. The ATA device allows flexible call routing with four sets of gateway parameters and configurable dial plans. The following table lists VoIP outbound call routing parameters.
6 Configuring the PSTN (FXO) Gateway on the SPA3102 Configuring VoIP Failover to PSTN The following table lists some examples. Example Description <9,:>xx.<:@gw1 Dial 9 to start outside dial tone, followed by one or more digits, and route the call to Gateway 1. [93]11<:@gw0> Route 911 and 311 calls to the local PSTN gateway <8,:1408>xxxxxxx<:@pstn. cisco.com:5061;usr=joe; pwd=joe_pwd;nat> Dial 8 to start outside dial tone, prepend 1408 followed by seven digits, and route the call to pstn.cisco.
Configuring the PSTN (FXO) Gateway on the SPA3102 Sharing One VoIP Account Between the FXS and PSTN Lines 6 Parameter Web Page Description Value Auto PSTN Fallback Line 1 If enabled, the ATA device automatically routes outbound calls to Gateway 0 when registration fails or network link is down. The default is yes. Sharing One VoIP Account Between the FXS and PSTN Lines Both the FXS (Line 1) and FXO (PSTN Line) can to receive incoming calls for a single VoIP account if they are different ports.
Configuring the PSTN (FXO) Gateway on the SPA3102 Other Options 6 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 • ”Call Progress Tones” section on page105 PSTN Call to Ring Line 1 This feature allows a PSTN caller to ring Line 1. When the PSTN line rings, the PSTN Line makes a local VoIP call to Line 1. If Line 1 is busy, it stops.
Configuring the PSTN (FXO) Gateway on the SPA3102 Call Scenarios 6 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: • number of frequency components • frequency and amplitude of each component • cadence information.
Configuring the PSTN (FXO) Gateway on the SPA3102 Call Scenarios 6 PSTN to VoIP Call with and Without Ring-Thru The PSTN caller calls the PSTN line connected to the FXO port. Ring-Thru is disabled. After the call rings for a delay equal to the value in PSTN Answer Delay, the VoIP gateway answers the call and prompts the PSTN caller to enter a PIN number (assuming PIN authentication is enabled). After a valid PIN is entered, the caller is prompted to dial the VoIP number.
Configuring the PSTN (FXO) Gateway on the SPA3102 Call Scenarios 6 The number dialed is processed by the dial plan corresponding to the VoIP caller. If the dial plan choice is 0, no dial plan is needed and the user hears the PSTN dial tone right after the PIN is entered. If the dial plan choice is not 0, the final number returned from the dial plan after the complete number is dialed by the caller is dialed to the PSTN.
6 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 or when the source IP address of the inbound INVITE matches one of the VoIP Access List patterns.
Configuring the PSTN (FXO) Gateway on the SPA3102 Call Scenarios 6 Call Forwarding to PSTN Gateway This section describes a number of scenarios that forward calls to the PSTN gateway.
Configuring the PSTN (FXO) Gateway on the SPA3102 Call Scenarios 6 Forward to a Particular PSTN Number In this scenario, the forward destination is set to @gw0>. 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. This is a special case of one-stage dialing where the target number is specified in the configuration.
A 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. NOTE This information applies to the SPA2102, SPA3102, and SPA8000 routers. To configure router settings for the PAP2T, WRP400, and WRTP54G, see the user guide for the router.
A 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. MAC Address MAC address of the ATA device. Client Certificate Status of the client certificate, which authenticates the ATA device for use in the ITSP network.
A 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. Broadcast Bytes Recv Total number of broadcast bytes received and processed. Broadcast Pkts Dropped Total number of broadcast packets received but not processed.
A 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. The default is 255.255.255.0. Gateway The default gateway used by ATA device when DHCP is disabled. The default is 0.0.0.0.
A 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 DNS servers. When DHCP is disabled, enter the primary DNS server. The default is 0.0.0.0.
A 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. If you do not wish to re-register the MAC address with your ISP, you may assign the MAC address you have currently registered with your ISP to the router with the MAC Address Clone feature.
A ATA Routing Field Reference LAN Setup page Maximum Uplink Speed The maximum bandwidth for LAN to WAN throughput. 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. VLAN ID The VLAN tag for the VLAN to which the ATA device is assigned. LAN Setup page You can use the LAN Setup page to enter your LAN settings.
A 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. Enable DHCP Server Options are Yes or No for the DHCP Server to provide an IP address. DHCP Lease Time Provided by the DHCP Server. IP renewal process begins when the time expires.
A 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. Service Name Any name to call the port forwarding starting port. Starting Port The starting port of the port range you wish to forward.
A 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.
B ATA Voice Field Reference This chapter describes the settings that you can configure under the Voice tab in the administration web server pages. NOTE For information about the Voice > Provisioning tab, see the SPA Provisioning Guide.
B ATA Voice Field Reference Info page Info page You can use the Voice tab > Info page to view information about the ATA device.
B 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. RTP Packets Sent Total number of RTP packets sent (including redundant packets). RTP Bytes Sent Total number of RTP bytes sent. RTP Packets Recv Total number of RTP packets received (including redundant packets).
B 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. This parameter is stored in long term memory and survives after reboot or power cycle. Call Back Active Indicates whether a call back request is in progress. Options are either Yes or No.
B ATA Voice Field Reference Info page ATA Administration Guide Call 1 and 2 Remote Hold Indicates whether the far end has placed the call on hold. Call 1 and 2 Callback Indicates whether the call was triggered by a call back request. Call 1 and 2 Peer Name Name of the internal phone. Call 1 and 2 Peer Phone Phone number of the internal phone. Call 1 and 2 Call Duration Duration of the call. Call 1 and 2 Packets Sent Number of packets sent.
B 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. Current Netmask Displays the network mask assigned to the ATA device. Current Gateway Displays the default router assigned to the ATA device.
B ATA Voice Field Reference Info page Last PSTN Reason for SPA hanging up the FXO port.
B ATA Voice Field Reference Info page VoIP State May take one of the following values: • • • • • PSTN State Idle Collecting PSTN Pin Invalid PSTN PIN PSTN Caller Accepted Connected to PSTN May take one of the following values: • • • • • Idle Collecting PSTN Pin Invalid PSTN PIN PSTN Caller Accepted Connected to PSTN VoIP Tone Indicates what tone is being played to the VoIP call leg. PSTN Tone Indicate what tone is being played to the PSTN call leg.
B ATA Voice Field Reference Info page VoIP Call Decode Latency Number of milliseconds for decoder latency. VoIP Call Jitter Number of milliseconds for receiver jitter. VoIP Call Round Trip Delay Number of milliseconds for delay. VoIP Call Packets Lost Number of packets lost. VoIP Call Packet Error Number of invalid packets received. VoIP Call Mapped RTP Port The port mapped for Real Time Protocol traffic for Call 1/2.
B ATA Voice Field Reference System page System page You can use the Voice tab > System page to configure your system and network connections.
B 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. NetMask The NetMask used by ATA device when DHCP is disabled. The default is 255.255.255.0. Gateway The default gateway used by ATA device when DHCP is disabled. The default is 0.0.0.0.
B 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. With parallel DNS query mode, the ATA device sends the same request to all the DNS servers at the same time when doing a DNS lookup, the first incoming reply is accepted by the ATA device.
B 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. SIP page You can use the Voice tab > SIP page to configure the SIP settings.
B ATA Voice Field Reference SIP page SIP User Agent Name SIP Server Name User-Agent header used in outbound requests. The default is $VERSION. If empty, the header is not included. Macro expansion of $A to $D corresponding to GPP_A to GPP_D allowed. Server header used in responses to inbound responses. The default is $VERSION. SIP Reg User Agent User-Agent name to be used in a REGISTER request. If this Name value is not specified, the SIP User Agent Name parameter is also used for the REGISTER request.
B ATA Voice Field Reference SIP page Escape Display Name Lets you keep the Display Name private. Select yes if you want the ATA device to enclose the string (configured in the Display Name) in a pair of double quotes for outbound SIP messages. Any occurrences of or \ in the string is escaped with \ and \\ inside the pair of double quotes. Otherwise, select no. The default is no. RFC 2543 Call Hold Configures the type of call hold: a:sendonly or 0.0.0.0. The default is no; do not use the 0.0.0.
B 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. The default is 32. SIP Timer D ACK hang-around time, which can range from 0 to 64 seconds. The default is 32.
B ATA Voice Field Reference SIP page Reg Retry Long Intvl When registration fails with a SIP response code that does 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. This value should be much larger than the Reg Retry Intvl value, which should not be 0. The default is 1200. Reg Retry Random Delay Random delay range (in seconds) to add to Register Retry Intvl when retrying REGISTER after a failure.
B 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. The default is 30. Voice tab > SIP page > RTP Parameters section RTP Port Min Minimum port number for RTP transmission and reception.
B 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). The last RTCP packet contains an additional BYE packet.
B ATA Voice Field Reference SIP page Voice tab > SIP page > SDP Payload Types section NSE Dynamic Payload NSE dynamic payload type. The valid range is 96-127. AVT Dynamic Payload AVT dynamic payload type. The valid range is 96-127. INFOREQ Dynamic Payload INFOREQ dynamic payload type. G726r16 Dynamic Payload G.726-16 dynamic payload type. The valid range is 96-127. G726r24 Dynamic Payload G.726-24 dynamic payload type. The valid range is 96-127. G726r40 Dynamic Payload G.
B ATA Voice Field Reference SIP page G726r32 Codec Name G.726-32 codec name used in SDP. G726r40 Codec Name G.726-40 codec name used in SDP. G729a Codec Name G.729a codec name used in SDP. G729b Codec Name G.729b codec name used in SDP. The default is G726-32. The default is G726-40. The default is G729a. The default is G729ab. G723 Codec Name G.723 codec name used in SDP. The default is G723. EncapRTP Codec Name EncapRTP codec name used in SDP. The default is EncapRTP.
B 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. Select yes or no from the drop-down menu. The default is no. Send Resp To Src Port Sends responses to the request source port instead of the VIA sent-by port.
B 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. NAT Keep Alive Intvl ATA Administration Guide Interval between NAT-mapping keep alive messages. The default is 15.
B 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 Option This feature controls which proxy debuy messages to log. The choices are as follows: • • • none—No logging. • 1-line excl. NTFY—Logs the start-line only for all messages except NOTIFY requests/responses. • 1-line excl.
B 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. The following options are available: • onhook only: An incoming call is directed to a specified trunk line only if the call state is onhook. • any state: An incoming call is directed to any specified trunk line without regard to the call state.
B 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). Second Dial Tone Alternative to the Dial Tone when the user dials a three-way call. The default is 420@-19,520@-19;10(*/0/1+2). Outside Dial Tone Alternative to the Dial Tone.
B ATA Voice Field Reference Regional page Ring Back 2 Tone Your ATA device plays this ringback tone instead of Ring 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 cadence is 1s on and 1s off. The default is 440@-19,480@-19;*(1/1/1+2). Confirm Tone Brief tone to notify the user that the last input value has been accepted. The default is 600@-16; 1(.25/.25/1).
B 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). Secure Call Indication Tone Played when a call has been successfully switched to secure mode.
B 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. The default is 60(.2/.4,.2/.4,.2/4). Ring7 Cadence Cadence script for distinctive ring 7. The default is 60(.4/.2,.4/.2,.4/4).
B 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). CWT9 Cadence Cadence script for distinctive CWT 9. This field is for the SPA2102 only. The default is 30(.3/9.7).
B 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. Ring9 Name Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 9 for the inbound call. This field is for the SPA2102 only. The default is Bellcore-r9.
B ATA Voice Field Reference Regional page Hook Flash Timer Max Maximum on-hook time before off-hook qualifies as hookflash. More than this the on-hook event is treated as onhook (no hook-flash event). Range: 0.4–1.6 seconds. The default is 0.9. Callee On Hook Delay Phone must be on-hook for at this time in sec before the ATA device will tear down the current inbound call. It does not apply to outbound calls. Range: 0–255 seconds. The default is 0.
B 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.
B 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. Call Back Deact Code Cancels a callback. Call Back Busy Act Code Starts a callback when the last outbound call is busy. This field is only found in the PAP2T. The default is *86.
B ATA Voice Field Reference Regional page Block Last Act Code Blocks the last inbound call. Block Last Deact Code Cancels blocking of the last inbound call. Accept Last Act Code Accepts the last outbound call. It lets the call ring through when do not disturb or call forwarding of all calls are enabled. The default is *60. The default is *80. The default is *64. Accept Last Deact Code Cancels the code to accept the last outbound call. CW Act Code Enables call waiting on all calls.
B 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. The default is *85. CWCID Act Code Enables call waiting, caller ID generation. The default is *25. CWCID Deact Code Disables call waiting, caller ID generation.
B 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. Modem pass-through mode can be triggered only by pre-dialing this code. FAX Line Toggle Code Toggles the line to a fax machine. This field is not found in the PAP2T. The default is #99.
B ATA Voice Field Reference Regional page Feature Dial Services Codes These codes tell the ATA device what to do when the user is listening to the first or second dial tone. One or more *code can be configured into this parameter, such as *72, or *72|*74|*67|*82, etc. Max total length is 79 chars. This parameter applies when the user has a dial tone (first or second dial tone).
B ATA Voice Field Reference Regional page Voice tab > Regional page > Vertical Service Announcement Codes section (SPA2102, SPA8000) Service Annc Base Number Base number for service announcements. Service Annc Extension Codes Extension codes for service announcements. Voice tab > Regional page > Outbound Call Codec Selection Codes section These codes automatically appended to the dial-plan. So no need to include them in dial-plan (although no harm to do so either).
B 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 Code Makes this codec the preferred codec for the associated call. The default is *0172616. Force G726r16 Code Makes this codec the only codec that can be used for the associated call. The default is *0272616. Prefer G726r24 Code Makes this codec the preferred codec for the associated call. The default is *0172624.
B ATA Voice Field Reference Regional page Voice tab > Regional page > Miscellaneous section Set Local Date (mm/dd) Sets the local date (mm stands for months and dd stands for days). The year is optional and uses two or four digits. Set Local Time (HH/ Sets the local time (hh stands for hours and mm stands for mm) minutes). Seconds are optional. Time Zone Selects the number of hours to add to GMT to generate the local time for caller ID generation.
B ATA Voice Field Reference Regional page Daylight Saving Time Rule Enter the rule for calculating daylight saving time; it should include the start, end, and save values. This rule is comprised of three fields. Each field is separated by ; (a semicolon) as shown below. Optional values inside [ ] (the brackets) are assumed to be 0 if they are not specified. Midnight is represented by 0:0:0 of the given date. SYNTAX: Start = ; end=; save = .
B ATA Voice Field Reference Regional page Daylight Saving Time Enable Daylight Saving Time can be turned on or off. This option 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. The range is 6.000 to -12.000. The default is -3. FXS Port Output Gain Output gain in dB, up to three decimal places. The range is 6.000 to -12.000.
B 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. • DTMF (Denmark)—CID only. DTMF sentbefore first ring with no polarity reversal and no DTAS. • ETSI DTMF—CID only.
B ATA Voice Field Reference Line page More Echo Suppression Enable or disable more echo suppresion. The default is no. This field is not found in the PAP2T. Line page Depending on the ATA device, there may be one or more Line pages (L1, L2, and so on). You can use the Voice tab > Line page to configure the lines for voice service.
B 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. Voice tab > Line page > Line Enable section Line Enable To enable this line for service, select yes. Otherwise, select no. The default is yes.
B ATA Voice Field Reference Line page SAS Inbound RTP Sink This setting works around devices that do not play inbound RTP if the streaming audio server line declares itself as a send-only device and tells the client not to stream out audio. Enter a Fully Qualified Domain Name (FQDN) or IP address of an RTP sink; this value is used by the streaming audio server line in the SDP of its 200 response to an inbound INVITE message from a client.
B ATA Voice Field Reference Line page NAT Keep Alive Dest Destination that should receive NAT keep alive messages. If the value is $PROXY, the messages are sent to the current proxy server or outbound proxy server. The default is $PROXY. Voice tab > Line page > Network Settings section SIP ToS/DiffServ Value TOS/DiffServ field value in UDP IP packets carrying a SIP message. The default is 0x68. SIP CoS Value [0-7] CoS value for SIP messages. The default is 3.
B 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. As a result, TCP overcomes the main disadvantages of UDP. In addition, for security reasons, most corporate firewalls block UDP ports.
B 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. This feature was requested by Bell Canada (Nortel) to limit the registration of SIP accounts. The default is yes.
B 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. To enable the Restrict Source IP feature, select yes. Otherwise, select no. If configured, the PAP2T will drop all packets sent to its SIP Ports originated from an untrusted IP address.
B ATA Voice Field Reference Line page Use Anonymous with RPID When set to yes, use “anonymous” in the SIP message 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 FROM The IP address of the local address enclosed in the FROM of the SIP message. This field is found on the SPA2102 only. Default is no.
B 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 Proxy Enablse the use of an Outbound Proxy. If set to no, the Outbound Proxy and Use OB Proxy in Dialog parameters are ignored. The default is no. Use OB Proxy In Dialog Whether to force SIP requests to be sent to the outbound proxy within a dialog.
B 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 Prefix If enabled, the PAP2T will automatically prepend the Proxy or Outbound Proxy name with _sip._udp when performing a DNS SRV lookup on that name. The default is no.
B 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. Call Capacity Maximum number of calls allowed on this line interface. Choices: {unlimited,1,2,3,…25 }. Default is 16.
B 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. Cfwd Busy Serv Enable Call Forward Busy Service The default is yes. Cfwd No Ans Serv Enable Call Forward No Answer Service The default is yes.
B ATA Voice Field Reference Line page Call Back Serv Enable Call Back Service. Three Way Call Serv Enable Three Way Calling Service. Three Way Calling is required for Three Way Conference and Attended Transfer. The default is yes. Three Way Conf Serv Enable Three Way Conference Service. Three Way Conference is required for Attended Transfer. The default is yes. Attn Transfer Serv Enable Attended Call Transfer Service. Three Way Conference is required for Attended Transfer. The default is yes.
B 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. So, if the G.729a codec is enabled and included in the codec list, that resource is tied up until the end of the call whether or not the call actually uses G.729a. If the G.729a resource is already allocated and since only one G.
B 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). The default is yes. Voice tab > Line page > Dial Plan section The default dial plan script for each line is as follows: (*xx|[3469]11|0|00|[29]xxxxxx|1xxx[2-9]xxxxxx|x xxxxxxxxxxx.).
B 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[29]xxxxxxS0|xxxxxxxxxxxx.) The dial plan syntax is expanded in the SPA3102 to allow the designation of three parameters to be used with a specific gateway: • • • uid – the authentication user-id pwd – the authentication password nat – if this parameter is present, use NAT mapping Each parameter is separated by a semi-colon (;).
B ATA Voice Field Reference Trunk Group page (SPA8000) Emergency Number Comma separated list of emergency number patterns. If 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. Maximum number length is 63 characters. The default is blank.
B 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 no. The default is yes. Voice tab > Trunk Group page > Network Settings section SIP ToS/DiffServ Value TOS/DiffServ field value in UDP IP packets carrying a SIP message. The default is 0x68. SIP CoS Value [0-7] CoS value for SIP messages. The default is 3.
B 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. Otherwise, select no. The default is no. Auth ResyncReboot If this feature is enabled, the ATA device authenticates the sender when it receives the NOTIFY resync reboot (RFC 2617) message.
B 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 excl. NTFY—Logs the start-line only for all messages except NOTIFY requests/responses. • 1-line excl. REG—Logs the start-line only for all messages except REGISTER requests/responses. • 1-line excl.
B 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. To enable the Restrict Source IP feature, select yes. Otherwise, select no. If configured, the PAP2T will drop all packets sent to its SIP Ports originated from an untrusted IP address.
B 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. Otherwise, select no to use the user ID and password. The default is no. Auth ID Authentication ID for SIP authentication.
B 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. If not, the SUA rejects the call with a 486 response. If there is spare capacity, the SUA consults the Contact List to determine which lines to ring (that is, for the proxy to send SIP INVITE to), and starts "hunting.
B 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. This Trunk Dial Plan typically is redundant since the trunk should trust the number sent by the Line SUA. By default the trunk dial plan allows any non-empty number: ([*#0-9AD][*#0-9A-D].
B 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 Proxy Enablse the use of an Outbound Proxy. If set to no, the Outbound Proxy and Use OB Proxy in Dialog parameters are ignored. The default is no. Outbound Proxy SIP Outbound Proxy Server where all outbound requests are sent as the first hop.
B 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 Prefix If enabled, the PAP2T will automatically prepend the Proxy or Outbound Proxy name with _sip._udp when performing a DNS SRV lookup on that name. The default is no.
B 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 • ”Dial Plans section” section on page 201 • ”VoIP-To-PSTN Gateway Setup section” section on page 202 • ”VoIP Users and Passwords (HTTP Authentication) section” section on page 204 • ”FXO (
B ATA Voice Field Reference PSTN Line page (SPA3102) NAT Keep Alive Enable To send the configured NAT keep alive message periodically, select yes. Otherwise, select no. The default is no. NAT Keep Alive Msg Enter the keep alive message that should be sent periodically to maintain the current NAT mapping. If the value is $NOTIFY, a NOTIFY message is sent. If the value is $REGISTER, a REGISTER message without contact is sent. Escape sequence of %xx is also accepted.
B ATA Voice Field Reference PSTN Line page (SPA3102) Network Jitter Level Determines how jitter buffer size is adjusted by the ATA 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. However, the starting jitter buffer size value is larger for higher jitter levels.
B ATA Voice Field Reference PSTN Line page (SPA3102) SIP Remote-PartyID 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. 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.
B 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. To enable the Restrict Source IP feature, select yes. Otherwise, select no. If configured, the PAP2T will drop all packets sent to its SIP Ports originated from an untrusted IP address.
B ATA Voice Field Reference PSTN Line page (SPA3102) Use Outbound Proxy Enable the use of Outbound Proxy. If set to no, the Outbound Proxy parameter and Use OB Proxy in Dialog is ignored. The default is no. Outbound Proxy SIP Outbound Proxy Server where all outbound requests are sent as the first hop. Use OB Proxy In Dialog Whether to force SIP requests to be sent to the outbound proxy within a dialog. Ignored if the Use Outbound Proxy parameter is no, or if the Outbound Proxy parameter is empty.
B 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. This parameter is useful only if the primary and backup proxy server list is provided to the PAP2T via DNS SRV record lookup on the server name.
B 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. So, if the G.729a codec is enabled and included in the codec list, that resource is tied up until the end of the call whether or not the call actually uses G.729a.
B 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 Enable To enable the echo canceller to adapt, select yes. Otherwise, select no. The default is yes. G726-16 Enable To enable the use of the G726 codec at 16 kbps, select yes. Otherwise, select no. The default is yes. Echo Supp Enable To enable the use of the echo suppressor, select yes. Otherwise, select no.
B ATA Voice Field Reference PSTN Line page (SPA3102) FAX Codec Symmetric To force the ATA device to use a symmetric codec during 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. Otherwise, select no. The default is yes. FAX Passthru Method Select the fax passthrough method: None, NSE, or ReINVITE. The default is NSE.
B 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 Mode This parameter has three possible values: caller or callee - SPA will detect FAX tone whether it is callee or caller caller only - SPA will detect FAX tone only if it is the caller callee only - SPA will detect FAX tone only if it is the callee The default is caller or callee.
B ATA Voice Field Reference PSTN Line page (SPA3102) Voice tab > PSTN Line page > VoIP-To-PSTN Gateway Setup section VoIP-To-PSTN Gateway Enable Enable or disable VoIP-To-PSTN Gateway functionality. VoIP Caller Authentication Method Method to be used to authenticate a VoIP Caller to access the PSTN gateway. Choose from {none, PIN, HTTP Digest. The default is yes. The default is none.
B ATA Voice Field Reference PSTN Line page (SPA3102) VoIP Caller ID Pattern A comma-separated list of caller number templates such 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. If this parameter is blank (not specified), all callers are considered for PSTN gateway service. For example: 1408*, 1512???1234.
B 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/ 5/6/7/8 Auth ID The first of 8 user-id’s that a VoIP Caller can use to 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. If the credentials are missing or incorrect, the SPA will challenge the caller with a 401 response).
B ATA Voice Field Reference PSTN Line page (SPA3102) VoIP User 1/2/3/4/ 5/6/7/8 DP Index of the dial plan in the dial plan pool to be used with VoIP User 1. The default is 1. VoIP User 1/2/3/4/ 5/6/7/8 Password The password to be used with VoIP User 1. The user assumes the identity of VoIP User 1 must therefore compute the credentials using this password, or the INVITE will be challenged with a 401 response The default is blank.
B ATA Voice Field Reference PSTN Line page (SPA3102) VoIP DLG Refresh Intvl Interval between (SIP) Dialog refresh messages sent by 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. The refresh message is a SIP ReINVITE and the VoIP peer must response with a 2xx response.
B ATA Voice Field Reference PSTN Line page (SPA3102) PSTN Hook Flash Len The length of the hook flash in seconds. During a PSTN-toVoIP gateway call, the ATA device processes the out-ofband hook flash signal sent from the VoIP peer through a hook-flash (momentary on-hook signal) on the FXO port. This allows the VoIP peer to initiate a three-way conference call and subsequent call transfer. The duration of the on-hook signal can be configured using this parameter. The default is 0.25.
B 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 the length specified in the Long Silence Duration parameter during a gateway call The default is yes. Min CPC Duration Specify the minimum duration of a low tip-and-ring voltage (below 1V) for the Gateway to recognize it as a CPC signal or PSTN line removal. The default is 0.2.
B 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. Restrictions: • 2 frequency components must be given. If single frequency is desired, the same frequency is used for both • The tone level value is not used. –30 (dBm) should be used for now.
B ATA Voice Field Reference PSTN Line page (SPA3102) Disconnect Tone continued Spain—425@-10; 10(0.2/0.2/1,0.2/0.2/1,0.2/0.6/1) — 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.
B ATA Voice Field Reference PSTN Line page (SPA3102) Voice tab > PSTN Line page > International Control (Settings) section FXO Port Impedance Desired impedance of the FXO Port. Choose from {600, 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. The Disconnect Tone Script and Impedance values for various countries follos: US—480@-30,620@-30;4(.25/.
B 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. Ring Frequency Max Specify the higher limit of the ring frequency used to detect the ring signal. The default is 100. PSTN To SPA Gain dB of digital gain (or attenuation if negative) to be applied to the signal sent from the PSTN side to the SPA. The range is -15 to 12.
B 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). The default value is 40v.
B 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 – forward the caller to use the PSTN gateway @gw0 – forward to caller to the PSTN number (dialed automatically by the SPlocalA through the PSTN gateway) The default is blank.
B ATA Voice Field Reference User page Voice tab > User page > Selective Call Forward Settings section Cfwd Sel1- 8 Caller Caller number pattern to trigger Call Forward Selective 1, 2, 3, 4, 5, 6, 7, or 8. The default is blank. Cfwd Sel1 - 8 Dest Forward number for Call Forward Selective 1, 2, 3, 4, 5, 6, 7, or 8. Same as Cfwd All Dest. The default is blank. Block Last Caller ID of caller blocked via the Block Last Caller service. The default is blank.
B 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.
B ATA Voice Field Reference User page Accept Media Loopback Request Controls how to handle incoming requests for loopback 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 manual —rings the phone first, and the call must be picked up manually before loopback starts. The default is Automatic.
B 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. Hold Reminder Ring Ring pattern for reminder of a holding call when the phone is on-hook. The default is None. Call Back Ring Ring pattern for call back notification. The default is None.
B ATA Voice Field Reference PSTN User page (SPA3102 Only) Ring On No New VM 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. The default is no. PSTN User page (SPA3102 Only) On the SPA3102, you can use the Voice tab > PSTN User page to configure the PSTN user settings.
B 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. Voice tab > PSTN User page > PSTN Ring Thru Line 1 Ring Settings section Default Ring The default ring to be used to ring through Line 1. Choose from {1,2,3,4,5,6,7,8,Follow Line 1}.
C 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). NOTE For instructions about provisioning, see the SPA Provisioning Guide in Cisco Partner Central, http://www.cisco.com/web/partners/sell/smb.
C Provisioning Reference (WRP400) Feature/XML Tag Parameters Examples RTSP rtsp_enable rtsp_enable: Real Time Streaming Protocol (RTSP); 1 (enabled) or 0 (disabled) To enable RTSP: rtsp_enable=1 IGMP force_igmp_version,multicast _pass,multicast_immediate_leave force_igmp_version: Specifies the version of IGMP that is supported; 1 (IGMP v1, RFC 1112), 2 (IGMP v2, RFC 2236) or 3 (IGMP v3, RFC 3376) To disable RTSP: rtsp_enable=0
C Provisioning Reference (WRP400) Feature/XML Tag Parameters Examples QoS Category Priority Rule category_ number,name, priority,port_range To configure a rule for an application: category_num= 1,name= ap1, priority=3,port_range= 111;222; 0;333;444;1 category_num: QoS Category number; 1 (application), 2 (online game), 3 (MAC address), 4 (Ethernet port) name: Name string, corresponding to the select
C Provisioning Reference (WRP400) Feature/XML Tag Parameters Examples Basic Wireless Settings for Primary Network wl_net_mode,wl _closed,wl_ssid wl_net_mode: Network mode; mixed, b-only, g-only, or disabled To enable SSID-1 and specify the SSID name: wl_net_mode =g-only,wl_closed=0, wl_ssid=aaabbb Basic Wireless Settings for Secondary or Guest Network wl_closed: SSID broadcast status; 1 (disab
C Provisioning Reference (WRP400) Feature/XML Tag Parameters Examples Wireless Security for SSID1 wl_security _mode2= [mode],[parameters] To disable Wireless Security 1: wl_security_mode2= disabled wl1_securit y_mode2= [mode],[parameters] To disable Wireless Security 2: wl1_security_mode2=di sabled Wireless Security f
C Provisioning Reference (WRP400) Feature/XML Tag Parameters Examples WPA Personal and WPA2 Personal Parameters To enable Wireless WPA Personal, specify the keys and set the renewal rate: wl_ security_ mode2= wpa_personal,wl_ crypto=aes, wl_wpa_ psk=personal, wl_wpa_gtk_ rekey=700 wl_crypto: WPA algorithms; tkip (TKIP) or aes (AES) wl_wpa_psk: WPA shared key; enter from 8 to 63 ASCII characters wl_wpa_gtk_rekey: WPA group key renewal; numerals from 600 to 720
C Provisioning Reference (WRP400) Feature/XML Tag Parameters Examples LAN DHCP dhcp_lease,dhcp_defa ult_lease To set the client lease time: dhcp_default_lease=888 dhcp_lease: Client lease time in minutes; numerals from 1 to 9999 To set lease time and default lease time: dhcp_lease=777,dhcp_ default_lease=888 dhcp_default_lease: Default lease time in minutes; numerals from 1 to 9999 NOTE: Dhcp_default_lease allows the Serv
C Provisioning Reference (WRP400) Feature/XML Tag Parameters WAN Type wan_proto=[mode], [parameters] Examples wan_proto: Internet connection type; dhcp, static, pppoe, pptp, l2tp, heartbeat DHCP Parameters No other settings are required. Static IP Parameters wan_ipaddr: WAN IP address wan_netmask: WAN subnet mask To configure a DHCP connection: wan_proto=dhcp To configure a Static IP connection: wan_proto=static,wan_ ipaddr=192.168.0.
C Provisioning Reference (WRP400) Feature/XML Tag Parameters Examples Heartbeat for Telstra Cable Network Parameters To configure a Telstra Cable connection: wan_proto= heartbeat,ppp_username=adc,ppp_ passwd=def,hb_ server_ip= 192.168. 0.
C Provisioning Reference (WRP400) Feature/XML Tag Parameters Examples Fail Pattern: ppp_demand=1,ppp_ idletime= 66666 ppp_demand=0,ppp_ redialperiod=777 ppp_demand=1 ppp_demand=0 ppp_demand=1,ppp_ redialperiod=77 ppp_demand=0,ppp_ idletime= 666 WAN Host wan_hostname=host_test, wan_domain=domain wan_hostname: WAN hostna
C Provisioning Reference (WRP400) Feature/XML Tag Parameters Examples Fail Pattern mtu_enable=0,wan_mtu= 999 wan_mtu=777 WAN DNS wan_dns wan_dns: DNS IP address; separate multiple addresses with a space WAN DNS, continued To specify one DNS address: wan_dns=192.168.0.21 To specify multiple DNS addresses: wan_dns=192.168.0.21 192.168.0.22 wan_dns=192.168.0.21 192.168.0.22 192.168.
C Provisioning Reference (WRP400) Feature/XML Tag Parameters Examples Single Port Forwarding forward _single=name:on|off:both|tcp|udp:external -port:internal-port:ip To forward FTP to 192.168.15.18: forward_singl e=FTP:on:tcp:21:21:18 NOTE: To configure port forwarding, you also should configure a DHCP reservation for the designated server.
C Provisioning Reference (WRP400) Feature/XML Tag Parameters Examples Port Range Forwarding forward _single=name:on|off:both|tcp|udp:port range start:port range end:ip To allow forwarding on two specified port ranges: forward_port=prf1:on:tcp:555:666:18 forward_port=prf2:on:both:777:888:19 NOTE: To configure port forwarding, you al
C Provisioning Reference (WRP400) Feature/XML Tag Parameters Examples Router Syslog log_provision To configure console display and system log: log_provision=2 ATA Administration Guide log_provision: Type of log; 0 (console display), 1 (system log), or 2 (console display and system log) 234
D Troubleshooting This appendix provides solutions to problems that may occur during the installation and operation of the ATA devices. NOTE If you can't find an answer here, visit www.cisco.com/go/smallbiz. Q. I want to use a different computer to access the administration web server. The address I entered did not work. A. Use the Interactive Voice Response Menu to find out the Internet IP address. Follow these steps: 1. Use a telephone connected to the Phone 1 port of the ATA device. 2.
D 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. Alternatively, from your browser you can select File > Save as > HTML from any of the administration web server pages. Do this in Admin, Advanced mode.
D 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. 2. Press 1 to confirm the operation. Press * to cancel the operation. 3.
Troubleshooting D NOTE STUN does not work with a symmetric NAT router. Enable debug through syslog (see FAQ#10), and set STUN Test Enable to yes. The messages indicate whether you have symmetric NAT or not.
E 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 ATA Administration Guide Device Dimensions 3.98” x 3.98” x 1.10” (101 x 101 x 28 mm) W x H x D Unit Weight 5.
E Environmental Specifications SPA2102 Storage Humidity 10% to 90% relative humidity, Non-Condensing Device Dimensions 3.98” x 3.98” x 1.10” (101 x 101 x 28 mm) W x H x D Unit Weight 5.29 oz (0.
E Environmental Specifications SPA8000 Certificatio n FCC (Part 15 Class B), CE, ICES-003, A-Tick Certification, RoH Operating Temp 32º to 113º F(0 to 45ºC) Storage Temp -13º to 185ºF (-25 to 85ºC) Operating Humidity 10% to 90% relative humidity, Non-Condensing Storage Humidity 10% to 90% relative humidity, Non-Condensing Device Dimensions 6.69” x 1.54” x 8.66” (170 x 39 x 220 mm) Unit Weight 2.85 lbs (1.
E Environmental Specifications WRP400 WRP400 Device Dimensions 5.51” x 5.51” x 1.06” (140 x 140 x 27 mm) Unit Weight 10.05 oz (285 g) Power External, Switching 5VDC 2A Certificatio n FCC (Part 15 Class B), CE, ICES-003, RoHS, UL, A-Tick, NZ Telepermit, CB, Wi-Fi (802.11b + WPA2, 802.
E Environmental Specifications WRTP54G ATA Administration Guide Operating Humidity 10% to 85% relative humidity, Non-Condensing Storage Humidity 5% to 90% relative humidity, Non-Condensing 243
F 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 244 • “Related Documentation,” on page 245 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. If you use an older web browser, you may have to add http:// in front of the web address.
F 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.
F Where to Go From Here Related Documentation Document Title Description Intended Audience SPA9x2 Phone User Guide • • • Phone setup VARs and phone endusers • Administration and use of Cisco Small Business ATAs • PAP2T, SPA2102, SPA3102, SPA8000, WRP400, and WRTP54G Analog Telephone Adapter Administration Guide Phone features SPA9x2 series IP phones VARs, system administrators, and Service Providers User Guide for switch User Guide for router ATA Administration Guide 246
G Additional Information This appendix provides links to resources that provide additional information about Cisco Small Business and Cisco Small Business Pro products and services. ATA Administration Guide Resource Location End User License Agreement www.cisco.com/go/smallbiz Regulatory Compliance and Safety Information www.cisco.com/go/smallbiz Warranty Information www.cisco.com/go/smallbiz Cisco Partner Central site for Small Business www.cisco.
H Support Contacts To obtain current support contact information for Cisco Small Business and Small Business Pro products, visit the following URL: www.cisco.
