Cisco SPA1001 Administration Guide

Cisco SPA1001 Manual

Get Cisco SPA1001 PDF manuals and user guides View all Cisco SPA1001 manuals
Add to My Manuals
Save this manual to your list of manuals

Cisco SPA1001 manual content summary:

  • Cisco SPA1001 | Administration Guide - Page 1
    Linksys ATA Administrator Users Guide Document Version 3.2 Corporate Headquarters Linksys 121 Theory Drive Irvine, CA 92617 USA http://www.linksys.com Tel: 949 823-1200 800 546-5797 Fax: 949 823-1100
  • Cisco SPA1001 | Administration Guide - Page 2
    ATA Administrator Guide Copyright ©2007 Cisco Systems, Inc. All rights reserved.Specifications are subject to change without notice. Linksys is a registered trademark or trademark of Cisco Systems, Inc. and/or its affiliates in the U.S. and certain other countries. Other brands and product names
  • Cisco SPA1001 | Administration Guide - Page 3
    Document Audience i-xi Linksys Analog Telephone Adapters i-xi How This Document is Organized i-xii Document Conventions i-xii Related Documentation i-xiii Technical Support i-xiii i-xiii 1 C H A P T E R Introducing Linksys Analog Telephone Adapters 1-1 Overview 1-1 Ensuring Voice Quality 1-3 Audio
  • Cisco SPA1001 | Administration Guide - Page 4
    12 SIP-NAT Interoperation 1-12 Where to Go From Here 1-13 Getting Started 2-1 Linksys Analog Telephone Adapters (ATAs) 2-1 Caring for Your Hardware 2-2 AG310 2-2 PAP2T 2-3 RTP300 2-4 SPA1001 2-5 SPA2102 2-6 SPA3102 2-7 SPA8000 2-9 WRP400 2-10 WRTP54G 2-11 WRT54GP2 2-13 Establishing Connectivity 2-14
  • Cisco SPA1001 | Administration Guide - Page 5
    with the SPA2102 or SPA8000 3-15 Managing Caller ID Service 3-17 Troubleshooting and Configuration FAQ 3-18 Contents 4 C H A P T E R Configuring the PSTN Gateway (FXO) 4-1 Overview 4-1 How VoIP-To-PSTN Calls Work 4-2 One-Stage Dialing 4-2 Document Version 3.1 Linksys ATA Administrator Guide v
  • Cisco SPA1001 | Administration Guide - Page 6
    Two-Stage Dialing 4-3 How PSTN-To-VoIP Calls Work 4-4 Terminating Gateway Calls 4-4 VoIP Outbound Call Routing 4-5 Configuring VoIP Failover to PSTN 4-7 Sharing One VoIP Account Between the FXS and PSTN Lines 4-7 Other Options 4-7 PSTN Call to Ring Line 1 4-8 Symmetric RTP 4-8 Call Progress Tones
  • Cisco SPA1001 | Administration Guide - Page 7
    Settings 5-35 Call Feature Settings 5-38 Proxy and Registration 5-38 Subscriber Information 5-40 Supplementary Service Subscription 5-40 Audio Configuration 5-42 Gateway Accounts (SPA3102/AG310) 5-45 VoIP Fallback to PSTN (SPA3102/AG310) 5-46 Dial Plan 5-46 FXS Port Polarity Configuration 5-47 PSTN
  • Cisco SPA1001 | Administration Guide - Page 8
    Contents Dial Plans 5-57 VoIP-To-PSTN Gateway Setup 5-57 VoIP Users and Passwords (HTTP Authentication) 5-59 Ring Settings 5-60 FXO (PSTN) Timer Values (sec) 5-60 PSTN Disconnect Detection 5-62 International Control (Settings) 5-63 User 1/2 Tab 5-65 Call Forward Settings 5-65 Selective Call Forward
  • Cisco SPA1001 | Administration Guide - Page 9
    Automatic Call Back C-9 Call FWD-Unconditional C-9 Call FWD - Busy C-10 Call FWD-No Answer C-11 Anonymous Call Blocking C-11 Distinctive/Priority Ringing and Call Waiting Tone C-12 Speed Calling-Up to Eight Numbers or IP Addresses C-12 Contents Document Version 3.1 Linksys ATA Administrator Guide
  • Cisco SPA1001 | Administration Guide - Page 10
    Contents Linksys ATA Administrator Guide x Document Version 3.1
  • Cisco SPA1001 | Administration Guide - Page 11
    provider for specific service parameters. Linksys Analog Telephone Adapters The following summarizes the ports and features provided by the Linksys ATAs described in this document. • PAP2T-Voice adapter with two FXS ports • SPA1001-Small VoIP adapter • SPA2102-Voice adapter with router Document
  • Cisco SPA1001 | Administration Guide - Page 12
    Linksys This chapter introduces the Linksys Analog Telephone Adapters Analog Telephone Adapters" (ATAs). Chapter 2, "Getting Started" This chapter describes how to use the different administration and configuration tools provided for managing Linksys ATAs. Chapter 3, "Configuring Linksys
  • Cisco SPA1001 | Administration Guide - Page 13
    SPA900 Series IP Phones Administrator Guide • Linksys Voice over IP Product Guide: SIP CPE for Massive Scale Deployment Technical Support If you are an end user of Linksys VoIP products and need technical support, contact the reseller or Internet telephony service provider (ITSP) that supplied the
  • Cisco SPA1001 | Administration Guide - Page 14
    Preface Linksys ATA Administrator Guide xiv Document Version 3.1
  • Cisco SPA1001 | Administration Guide - Page 15
    router Voice adapter with router and PSTN connectivity Voice adapter with support for up to eight FXS devices IP router with two FXS ports Wireless-G IP router with two FXS ports One (1) One (1) ADSL2+ gateway with VoIP and PSTN connectivity Document Version 3.1 Linksys ATA Administrator Guide
  • Cisco SPA1001 | Administration Guide - Page 16
    Overview Chapter 1 Introducing Linksys Analog Telephone Adapters Figure 1-1 illustrates how the different ATAs provide voice connectivity in a VoIP network, including the SPA3102, which acts as a SIP-PSTN gateway. As shown, the following devices also provide QoS-enabled IP routers in addition to
  • Cisco SPA1001 | Administration Guide - Page 17
    1 Introducing Linksys Analog Telephone Adapters Ensuring Voice Quality Each Linksys ATA is an intelligent low-density Voice over IP (VoIP) gateway that enables carrier-class residential and business IP Telephony services delivered over broadband or high-speed Internet connections. Linksys ATAs
  • Cisco SPA1001 | Administration Guide - Page 18
    Ensuring Voice Quality Chapter 1 Introducing Linksys Analog Telephone Adapters Audio Compression Algorithm Speech signals are sampled, quantized, and compressed before they are packetized and transmitted to the other end. For IP Telephony, speech signals are usually sampled at 8000 samples per
  • Cisco SPA1001 | Administration Guide - Page 19
    IP network. Reasonable delay figure should be about 50-100 ms. End-to-end delay larger than 300 ms is unacceptable to most callers. The Linksys ATA supports end-to-end delays well within acceptable thresholds. Feature Descriptions The Linksys ATA is a full featured, fully programmable phone adapter
  • Cisco SPA1001 | Administration Guide - Page 20
    of the Linksys ATA FXS ports and use it as a streaming audio source device. The corresponding Line (1 or 2) can be configured as a streaming audio server (SAS) such that when the Line is called, the Linksys ATA answers the call automatically and Linksys ATA Administrator Guide 1-6 Document Version
  • Cisco SPA1001 | Administration Guide - Page 21
    1 Introducing Linksys Analog Telephone Adapters Feature Descriptions starts streaming audio to the calling party provided the FXS port is off-hook. If the FXS port is on-hook when the incoming call arrives, the Linksys ATA replies with a SIP 503 response code to indicate "Service Not Available
  • Cisco SPA1001 | Administration Guide - Page 22
    Feature Descriptions Chapter 1 Introducing Linksys Analog Telephone Adapters The Linksys ATA has a Network Jitter Level control setting for each line of service. The jitter level decides how aggressively the Linksys ATA tries to shrink the jitter buffer over time to achieve a lower overall delay.
  • Cisco SPA1001 | Administration Guide - Page 23
    Chapter 1 Introducing Linksys Analog Telephone Adapters Technology Background Table 1-4 Linksys ATA Features Feature Signaling Hook Flash Event Configurable Dial Plan with Interdigit Timers Polarity Control Calling Party Control Report Generation and Event Logging Syslog and Debug Server Records
  • Cisco SPA1001 | Administration Guide - Page 24
    Technology Background Chapter 1 Introducing Linksys Analog Telephone Adapters Session Initiation Protocol Linksys ATAs are implemented using open standards, such as Session Initiation Protocol (SIP), allowing interoperation with all ITSPs supporting SIP. Figure 1-2 illustrates a SIP request for
  • Cisco SPA1001 | Administration Guide - Page 25
    Chapter 1 Introducing Linksys Analog Telephone Adapters Technology Background A typical application of a NAT is to allow all the devices in a subscriber home network to access the Internet through a router with a single public IP address assigned by an ISP. The IP header of the packets sent from
  • Cisco SPA1001 | Administration Guide - Page 26
    Linksys Analog Telephone Adapters With symmetric NAT all requests from the same internal IP address and port to a specific destination IP address and port are mapped to a unique external source IP address following website: http://www.voip-info.org/wiki-Open+Source+VOIP+Software STUN does not work
  • Cisco SPA1001 | Administration Guide - Page 27
    1 Introducing Linksys Analog Telephone Adapters Where to Go From Here responds to a special NAT-Mapping-Discovery request by sending back a message to the source IP address/port of the request, where the message contains the source IP address/port of the original request. The Linksys ATA system
  • Cisco SPA1001 | Administration Guide - Page 28
    Where to Go From Here Chapter 1 Introducing Linksys Analog Telephone Adapters • SPA900 Series IP Phones Administrator Guide • Linksys Voice over IP Product Guide: SIP CPE for Massive Scale Deployment 1-14 Linksys ATA Administrator Guide Document Version 3.1
  • Cisco SPA1001 | Administration Guide - Page 29
    to the user guide for the specific device. For information about configuring the Public Switch Telephone Network (PSTN) connectivity options for the SPA3102 and AG310 refer to Chapter 3, "Configuring Linksys ATAs." Linksys Analog Telephone Adapters (ATAs) Linksys ATAs convert voice traffic into
  • Cisco SPA1001 | Administration Guide - Page 30
    Linksys Analog Telephone Adapters (ATAs) • WRTP54G, page 2-11 • WRT54GP2, page 2-13 • RTP300, page 2-4 Chapter 2 Getting Started Caring for Your Hardware AG310 The Linksys are made to the software configuration of the SPA3102, the Linksys ATA Administrator Guide 2-2 Document Version 3.1
  • Cisco SPA1001 | Administration Guide - Page 31
    Getting Started Linksys Analog Telephone Adapters (ATAs) Table 1-5 AG310 Front Panel LED Power Ethernet 1-4 Voice Status Phone Line DSL Internet Function Steady green indicates power on and Internet connection. Flashing indicates not connected to the Internet, booting or firmware upgrade. Steady
  • Cisco SPA1001 | Administration Guide - Page 32
    or firmware upgrade. Table 1-8 PAP2T Back Panel Port Phone 1/2 Ethernet Power Function Connect to an analog telephone or fax machine with an RJ-11 cable. Connect to a router or broadband (cable/DSL) modem. Connect to the 5-volt DC power adapter. The RTP300 provides an ATA with two FXS ports
  • Cisco SPA1001 | Administration Guide - Page 33
    cable. Connect to an analog telephone or fax machine using an RJ-11 cable. Connect to local IP devices, such as PCs, using an Ethernet cable. Connect to a 12v DC power supply. SPA1001 The SPA1001 provides one FXS port (see Figure 1-7). Document Version 3.1 Linksys ATA Administrator Guide 2-5
  • Cisco SPA1001 | Administration Guide - Page 34
    Linksys Analog Telephone Adapters (ATAs) Figure 1-7 SPA1001 Chapter 2 Getting Started SPA2102 The following table describes the LEDS and ports on the back panel of the device. Table 1-11 SPA1001 Back Panel LED/Port Phone Power Act Ethernet Function Connect to an analog telephone or fax machine
  • Cisco SPA1001 | Administration Guide - Page 35
    Started Figure 1-8 SPA2102 Linksys Analog Telephone Adapters (ATAs) SPA3102 The following tables describe the LEDS on the front panel and the ports on the back panel of the device. Table 1-12 SPA2102 Front Panel LED Power Internet Phone 1/2 Function Steady green indicates power on and Internet
  • Cisco SPA1001 | Administration Guide - Page 36
    telephone wall jack with an RJ-11cable. Connect to an analog telephone or fax machine with an RJ-11 cable. Connect to a network device, such as a PC or a switch with an Ethernet cable. Connect to a router or broadband (cable/DSL) modem. Connect to the power supply. Linksys ATA Administrator Guide
  • Cisco SPA1001 | Administration Guide - Page 37
    Chapter 2 Getting Started Linksys Analog Telephone Adapters (ATAs) SPA8000 The SPA8000 is an analog telephone adapter that supports connecting up to eight analog telephone devices, including up to four fax devices. The SPA8000 consists of four hardware modules, each of which is similar in
  • Cisco SPA1001 | Administration Guide - Page 38
    This 12-volt power adapter is to be used only with the SPA8000. Using it with any other Linksys product may damage the unit. WRP400 The WRP400 provides an ATA with two FXS ports and a Wireless-G multiport IP router (see Figure 1-11). The WRP400 provides connectivity to an analog telephone as well
  • Cisco SPA1001 | Administration Guide - Page 39
    12-volt power supply. The WRTP54G provides an ATA with two FXS ports and a Wireless-G multiport IP router (see Figure 1-12). The WRTP54G provides connectivity to an analog telephone as well as Internet connectivity to a LAN with a built-in four-port switch. The IP router provides QoS support, an
  • Cisco SPA1001 | Administration Guide - Page 40
    1-4 Power Function Connect to the broadband (cable/DSL) modem using an Ethernet cable. Connect to an analog telephone or fax machine using an RJ-11 cable. Connect to local IP devices, such as PCs, using an Ethernet cable. Connect to the 12-volt power supply. 2-12 Linksys ATA Administrator Guide
  • Cisco SPA1001 | Administration Guide - Page 41
    Chapter 2 Getting Started Linksys Analog Telephone Adapters (ATAs) WRT54GP2 The WRP54GP2 provides an ATA with two FXS ports and a Wireless-G multiport IP router (see Figure 1-13). The WRP54GP2 provides connectivity to an analog telephone as well as Internet connectivity to a LAN with a built-in
  • Cisco SPA1001 | Administration Guide - Page 42
    the power supply. Establishing Connectivity This section describes how to connect the Linksys ATA hardware. It includes the following topics: • Bandwidth Requirements, page 2-14 • Making the Physical Connections, page 2-15 Bandwidth Requirements Depending on how you have your IP phones configured
  • Cisco SPA1001 | Administration Guide - Page 43
    you have the following package contents: • Linksys phone adapter unit • Ethernet cable • RJ-11 phone cable (SPA3102/AG310 Only) • SPA Quickstart Guide • Volt power adapter You also need the following: • One or two analog touchtone telephones (or fax Machine) • Access to an IP network via an Ethernet
  • Cisco SPA1001 | Administration Guide - Page 44
    include remote provisioning, firmware upgrades, the administration web server, IVR functions, and SIP NOTIFY control messages (over Line 1 or 2). The IVR functions are accessed by connecting an analog telephone to Line 1 or Line 2. 2-16 Linksys ATA Administrator Guide Document Version 3.1
  • Cisco SPA1001 | Administration Guide - Page 45
    the IVR, connect an analog telephone to the Phone 1 or Phone 2 port. To access the administration web server, connect a PC to the AUX port. Note In the illustration, one fax machine is connected to each pair of ports to illustrate that only one T.38 connection is supported by each of the four
  • Cisco SPA1001 | Administration Guide - Page 46
    that provides hairpinning support to prevent this problem. Using the Interactive Voice Response Interface This section describes how to read or write basic network configuration settings using a touchtone telephone connected to one of the FXS (RJ-11) phone ports of the Linksys ATA. It includes
  • Cisco SPA1001 | Administration Guide - Page 47
    , "Linksys configuration menu. Please enter option followed by the pound key or hang-up to exit." None None IVR announces if DHCP in enabled or disabled. Enter 1 to enable Requires password Enter 0 to disable None IVR announces the current IP address of the WAN port. Enter IP address using
  • Cisco SPA1001 | Administration Guide - Page 48
    . Requires password None IVR announces the MAC address of Linksys ATA in hex string format. None IVR announces the version of the firmware running on the Linksys ATA. None IVR announces the current setting in the parameter. Enter IP address using numbers on the telephone key pad
  • Cisco SPA1001 | Administration Guide - Page 49
    Unit reboots and all configuration parameters are reset to factory default values. Note The items marked with "Requires Password" only require a password if the Administrator password is set. Entering a Password through the IVR To input the password using the phone keypad, the following translation
  • Cisco SPA1001 | Administration Guide - Page 50
    Connect an analog telephone to the Phone 1 or Phone 2 port on the Linksys ATA. b. Press **** on the keypad to access the IVR menu. c. Press 110# to determine the Internet (WAN) IP address. Direct the browser to the IP address of the Linksys ATA. Changing between the tabs on the Voice page does not
  • Cisco SPA1001 | Administration Guide - Page 51
    . When changes are saved, the Linksys ATA may reboot. Administrator Account Privileges The Linksys ATA supports two levels of administration privileges: Administrator and User. Both privileges can be password protected. Note By default, there are no passwords assigned for either the Administrator
  • Cisco SPA1001 | Administration Guide - Page 52
    Using the Administration Web Server Chapter 2 Getting Started 2-24 Linksys ATA Administrator Guide Document Version 3.1
  • Cisco SPA1001 | Administration Guide - Page 53
    Provisioning, page 3-4 • Configuring a Dial Plan, page 3-5 • Secure Call Implementation, page 3-10 • Configuring a Streaming Audio Server, page 3-12 • Using a FAX Machine with the SPA2102 or SPA8000, page 3-15 • Managing Caller ID Service, page 3-17 • Troubleshooting and Configuration FAQ, page 3-18
  • Cisco SPA1001 | Administration Guide - Page 54
    Type drop-down menu. This is the correct setting for most DSL users. b. Enter the values provided by the ITSP in the following fields: • PPPoE Login Name • PPPoE Login Password • PPPoE Service Name Configure the Linksys ATA parameters on the Line tab with the appropriate values for the specific
  • Cisco SPA1001 | Administration Guide - Page 55
    upgrade the Linksys ATA 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 no port is specified, the default port is used (69
  • Cisco SPA1001 | Administration Guide - Page 56
    HTTP (firmware upgrades using HTTPS are not supported). Remote upgrades are controlled by configuring the desired firmware image URL into the Linksys ATA via a remote profile resync. For further information about remote provisioning refer to the Linksys SPA Provisioning Guide. Configuration Profile
  • Cisco SPA1001 | Administration Guide - Page 57
    SPC. Configuring a Dial Plan The Linksys ATA allows each line to be configured with a distinct dial plan. The dial plan specifies how to interpret digit sequences dialed by the user, and how to convert those sequences into an outbound dial string. The Linksys ATA syntax for the dial plan closely
  • Cisco SPA1001 | Administration Guide - Page 58
    the SPA2102 supported dual-line telephones on a single FXS port. A subsequent hardware change revised the FXS ports on the SPA2102 and this function is to be reserved for future development. Dial Plan Rules This section describes the rules that apply to configuring and interpreting dial plans. Note
  • Cisco SPA1001 | Administration Guide - Page 59
    Configuring Linksys ATAs Configuring a Dial Plan dialed line" dial tone outside line" dial tone after the user presses 9, dial plan (affecting a specific line) by preceding the entire plan for the line associated with this dial plan. And, "L:8,S:4,( . . . before continuing to dial digits, in order
  • Cisco SPA1001 | Administration Guide - Page 60
    a hot line phone, which automatically calls 1 212 5551234. ( S0 ) The following provides a warm line to a local office operator (1000) after five seconds, unless a four-digit extension is dialed by the user. ( P5 | xxxx ) Linksys ATA Administrator Guide 3-8 Document Version
  • Cisco SPA1001 | Administration Guide - Page 61
    Chapter 3 Configuring Linksys ATAs Configuring a Dial Plan Dial Plan Timers The dial plan functionality is regulated by the following configurable parameters: • Interdigit_Long_Timer • Interdigit_Short_Timer • Dial_Plan ([1] and [2]) Interdigit Long Timer ParName Default Interdigit_Long_Timer
  • Cisco SPA1001 | Administration Guide - Page 62
    not expired, and it has a valid signature. The Linksys ATA can be configured so that, by default, all outbound calls are either secure or not secure. If secure by default, the user has the option to disable security when making a call by dialing *19 before dialing the target number. If not secure by
  • Cisco SPA1001 | Administration Guide - Page 63
    INFO messages to pass through with the message body unmodified. Generating a Mini-Certificate Linksys provides a configuration tool called gen_mc for the generation of MC and private keys with the following syntax: gen_mc ca-key user-name user-id expire-date Where: • ca-key is a text file with the
  • Cisco SPA1001 | Administration Guide - Page 64
    still receive audio while the call is holding, the Linksys ATA can be configured to contact an auto-answering streaming audio server (SAS) to stream audio to the holding party. When used this way, the SAS is referred to as an MOH Server. 3-12 Linksys ATA Administrator Guide Document Version 3.1
  • Cisco SPA1001 | Administration Guide - Page 65
    installing the music source using the media signal adapter and completing the required configuration on the Linksys ATA , when the line is called and the FXS port is off hook, the Linksys ATA answers the call automatically and streams audio to the caller. If the FXS port is on-hook when the incoming
  • Cisco SPA1001 | Administration Guide - Page 66
    with a Linksys ATA Line Configured as an SAS SPA1: IP=192.168.2.100 UserID[1]=1001, SIP Port[1]=5060 UserID[1]=1002, SIP Port[1]=5061 SPA2: IP=192.168.2.200 UserID[1]=2001, SIP Port[1]=5060 UserID[1]=2002, SIP Port[1]=5061 Phone 1 Phone 2 IP Network Phone 1 Phone 2 Media signal adapter Line
  • Cisco SPA1001 | Administration Guide - Page 67
    the User ID configured for the line attached to the audio source. If the line is not registered with the SIP proxy, enter the IP address and SIP port number configured for the line. Click Submit All Changes. Using a FAX Machine with the SPA2102 or SPA8000 Note T.38 Fax is only supported on the
  • Cisco SPA1001 | Administration Guide - Page 68
    use the following settings: Make sure the Cisco gateway is correctly configured for T.38 with the SPA dial peer. For example: fax protocol T38 fax rate voice fax-relay ecm disable fax nsf 000000 no vad Troubleshooting If have problems sending or receiving faxes, complete the following steps: Step
  • Cisco SPA1001 | Administration Guide - Page 69
    . For instructions, refer to Item 4. in "Troubleshooting and Configuration FAQ" at the end of this chapter. Identify the type of Fax machine connected to the ATA. Contact technical support. If you are an end user of Linksys VoIP products, contact the reseller or Internet telephony service provider
  • Cisco SPA1001 | Administration Guide - Page 70
    Use the Interactive Voice Response Menu to find out the Linksys ATA Internet IP address. Follow these steps: A. Use a telephone connected to the Phone 1 port of the Linksys ATA. B. Press **** (in other words, press the star key four times). C. Wait until you hear "Linksys configuration menu. Please
  • Cisco SPA1001 | Administration Guide - Page 71
    User or Admin account without a password and reconfigure the unit as necessary. 6. My Linksys ATA is behind a NAT device or firewall and I'm unable to make a call or I'm only receiving a one-way connection. What should I do? A. Configure your router to port forward "TCP port 80" to the ip address
  • Cisco SPA1001 | Administration Guide - Page 72
    to which the Linksys ATA is registering, if it supports NAT, using the parameter. D. Add a STUN server to allow traversal of UDP packets through the NAT device. On the SIP tab of the administration web server, set to yes, and enter the IP address of the STUN server
  • Cisco SPA1001 | Administration Guide - Page 73
    analog telephone or fax machine, configured using the Line tab • FXO port (Line)-Connected to a standard telephone wall jack for connectivity to the PSTN, configured using the PSTN Line tab Line 1 does not provide a gateway because it provides only VoIP service. The VoIP-To-PSTN calling function
  • Cisco SPA1001 | Administration Guide - Page 74
    the call is answered How VoIP-To-PSTN Calls Work To obtain PSTN services through the SPA3102, the VoIP caller establishes a connection with the PSTN Line by way of a standard SIP INVITE request addressed to the PSTN Line. The PSTN Line can be configured to support one-stage and two-stage dialing as
  • Cisco SPA1001 | Administration Guide - Page 75
    username value. Values 31-character string The password value. 31-character string Specifies the dial plan to be used for this VoIP Choice of 0-8 user. If 0, dial plan processing is disabled; the given target number is dialed to the PSTN as is. Note If Authentication is disabled, a default dial
  • Cisco SPA1001 | Administration Guide - Page 76
    PIN number. Note If Authentication is disabled, a default dial plan is used for all PSTN callers. Terminating Gateway Calls There are two call legs in a PSTN gateway call: the PSTN call leg and the VoIP call leg. A gateway call is terminated when either call leg is ended. The SPA3102 takes the FXO
  • Cisco SPA1001 | Administration Guide - Page 77
    the configured Line 1 service provider, by default. You can override this behavior by IP dialing, through which the calls can be routed to any IP address entered by the user. SPA3102 allows flexible call routing with four sets of gateway parameters and configurable dial plans. Table 1-30 lists VoIP
  • Cisco SPA1001 | Administration Guide - Page 78
    4 Configuring the PSTN Gateway (FXO) Table 1-30 VoIP Outbound Call Routing Parameters Parameters Group Gateway 1 Line 1 GW1 Nat Mapping Enable GW1 User ID GW2 Password Similar for GW 2, 3, and 4 Line 1 Line 1 Line 1 Line 1 Description Values Fully qualified domain name (or IP address) of
  • Cisco SPA1001 | Administration Guide - Page 79
    Configuring VoIP Failover to PSTN Configuring VoIP Failover to PSTN When power is disconnected from the SPA3102, the FXS port is connected to the FXO port. In this case, the telephone attached to the FXS port is electrically connected to the PSTN service with Call-Forward- automatically dialed by
  • Cisco SPA1001 | Administration Guide - Page 80
    caller and processes the number using the dial plan. The SPA3102 plays the to prompt the VoIP caller to enter the PSTN number. This tone should be specified to sound different from the PSTN dial tone. Call Scenarios This section describes some typical scenarios where the SPA3102
  • Cisco SPA1001 | Administration Guide - Page 81
    can be configured into the SPA3102 to automatically grant access to the PSTN caller without entering the PIN. In this case, the default PSTN dial plan is also used. The same scenario can be implemented using Ring-Thru. When the PSTN line rings, Line 1 rings also. This feature is called Ring-Thru
  • Cisco SPA1001 | Administration Guide - Page 82
    method, even when the source IP address of the INVITE request is in the . Using HTTP Digest Authentication The same scenario can be implemented with HTTP digest authentication when the calling device supports the configuration of a auth-ID and password to access the SPA3102 PSTN
  • Cisco SPA1001 | Administration Guide - Page 83
    Local Calls To implement this scenario, add the rule "xxxxxxx" to the Line 1 dial plan. When user dials 9, SPA3102 plays outside dial tone. The user then dials seven digits and the SPA3102 prepends 1408 before dialing the final number on the PSTN line. Document Version 3.1 Linksys
  • Cisco SPA1001 | Administration Guide - Page 84
    , add the rule "[39]11" to Line 1. When the user dials 311 or 911, the call is routed to the PSTN gateway. Note If the PSTN Line is busy after the user dials 311 or 911, the call still fails. For true life-line supports, therefore, the PSTN Line cannot be shared. Auto-Fallback to the
  • Cisco SPA1001 | Administration Guide - Page 85
    Tab, page 5-33 • PSTN Line Tab, page 5-49 • User 1/2 Tab, page 5-65 • PSTN User Tab (SPA3102/AG310), page 5-70 Note Througout this chapter, Linksys router. For information about the Provisioning page, see the Linksys SPA Provisioning Guide. Document Version 3.2 Linksys IP Phone Administrator Guide
  • Cisco SPA1001 | Administration Guide - Page 86
    current IP address assigned to the Linksys IP phone. Displays the current IP address assigned to the Linksys IP phone. Displays the network domain name of the Linksys IP phone. Displays the network mask assigned to the Linksys IP phone. Displays the default router assigned to the Linksys IP phone
  • Cisco SPA1001 | Administration Guide - Page 87
    that the phone is ready for use; Off indicates that the phone is in use. Indicates if the line has registered with the SIP proxy. Last date and time the line was registered. Number of seconds before the next registration renewal. Document Version 3.2 Linksys IP Phone Administrator Guide 5-3
  • Cisco SPA1001 | Administration Guide - Page 88
    manually modify it to clear or set the flag. Setting this value to Yes can activate stutter tone and VMWI signal. This parameter is stored in long term memory and survives after reboot or power cycle. Indicates whether a call Call 1/2. Linksys IP Phone Administrator Guide 5-4 Document Version 3.2
  • Cisco SPA1001 | Administration Guide - Page 89
    Dialing Timeout • PSTN Gateway Call Timeout • VoIP Gateway Call Timeout PSTN Activity Timer Shows the time (ms) before the SPA disconnects the current gateway unless the PSTN side has some audio activity. Mapped SIP Port Same as Line 1. Document Version 3.2 Linksys IP Phone Administrator Guide
  • Cisco SPA1001 | Administration Guide - Page 90
    VoIP call leg). Same as Line 1 Call 1. Same as Line 1 Call 1. Same as Line 1 Call 1. Same as Line 1 Call 1. Same as Line 1 Call 1. Same as Line 1 Call 1. Same as Line 1 Call 1. Same as Line 1 Call 1. Same as Line 1 Call 1. Same as Line 1 Call 1. Linksys IP Phone Administrator Guide 5-6 Document
  • Cisco SPA1001 | Administration Guide - Page 91
    Chapter 5 Linksys ATA Field Reference VoIP Call Packets Lost VoIP Call Packet Error VoIP Call Mapped RTP Port Same as Line 1 Call 1. Same as Line 1 Call 1. Same as Line 1 Call 1. Info Tab Document Version 3.2 Linksys IP Phone Administrator Guide 5-7
  • Cisco SPA1001 | Administration Guide - Page 92
    . The default is yes. Password for the administrator. The default is no password. Password for the user. The default is no password. Internet Connection Type (PAP2T) Field DHCP Static IP NetMask Description Enable or disable DHCP. The default is yes. Static IP address of Linksys IP phone, which
  • Cisco SPA1001 | Administration Guide - Page 93
    is generated. Zero (0) means no debug information is generated. To log SIP messages, Debug Level must be set to at least 2. The default is 0. IP address or name of primary NTP server. IP address or name of secondary NTP server. Document Version 3.2 Linksys IP Phone Administrator Guide 5-9
  • Cisco SPA1001 | Administration Guide - Page 94
    Linksys ATA Field Reference Field Syslog Server Debug Server Debug Level Description Specifies the IP address of the syslog server. Specifies the IP address . The default is 0, which indicates that no debug information is generated. 5-10 Linksys IP Phone Administrator Guide Document Version
  • Cisco SPA1001 | Administration Guide - Page 95
    be used in a REGISTER request. If this is not specified, the is also used for the REGISTER request. The default is blank. Accept-Language header used. There is no default (this indicates Linksys IP phone does not include this header). If empty, the header is not included. MIME
  • Cisco SPA1001 | Administration Guide - Page 96
    (if defined by RFC 261) if parameter is set to yes. The default is no. Lets you keep the Display Name private. Select yes if you want the Linksys IP phone to enclose the string (configured in the Display Name) in a pair of double quotes for outbound SIP messages. Any occurrences
  • Cisco SPA1001 | Administration Guide - Page 97
    is enabled, is added on top of the exponential back-off adjusted delay value. This feature was added in Release 5.1. The default value is 0, which disables the exponential back-off feature. Document Version 3.2 Linksys IP Phone Administrator Guide 5-13
  • Cisco SPA1001 | Administration Guide - Page 98
    default is 0.030. Number of successive ICMP errors allowed when transmitting RTP packets to the peer before the Linksys IP phone terminates the call. If value is set to 0, the Linksys IP phone ignores the limit on ICMP errors. The default is 0. 5-14 Linksys IP Phone Administrator Guide Document
  • Cisco SPA1001 | Administration Guide - Page 99
    the peer, it attempts to compute the round trip delay and show it as the value (ms) in the Info section of Linksys IP phone web page. The default is 0. Select yes if you want the Linksys IP phone to calculate the UDP header checksum for SIP messages. Otherwise, select no. The
  • Cisco SPA1001 | Administration Guide - Page 100
    SDP. The default is G729a. G.729b codec name used in SDP. The default is G729ab. G.723 codec name used in SDP. The default is G723. EncapRTP codec name used in SDP. The default is EncapRTP. NAT Support Parameters Field Description 5-16 Linksys IP Phone Administrator Guide Document Version 3.2
  • Cisco SPA1001 | Administration Guide - Page 101
    when it powers on. It contacts the configured STUN server, and the result of the discovery is reported in a Warning header in all subsequent REGISTER requests. If the Linksys IP phone detects symmetric NAT or a symmetric firewall, NAT mapping is disabled. The default is no. IP address or fully
  • Cisco SPA1001 | Administration Guide - Page 102
    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. Interval between NAT-mapping keep alive messages. The default is 15. 5-18 Linksys IP Phone Administrator Guide Document Version 3.2
  • Cisco SPA1001 | Administration Guide - Page 103
    number, as opposed to an internal extension. It is triggered by a, (comma) character encountered in the dial plan. The default is 420@-19;10(*/0/1). Prompts the user to enter a call forwarding phone number. The default is 520@-19,620@-19;10(*/0/1+2). Played when a 486 RSC is received for an outbound
  • Cisco SPA1001 | Administration Guide - Page 104
    Cfwd Dial Tone Holding Tone Conference Tone Secure Call Indication Tone VoIP PIN Tone PSTN PIN Tone Feature Invocation Tone Chapter 5 Linksys ATA Field Reference Played during an outbound call when the far end is ringing. The default is 440@-19,480@-19;*(2/4/1+2). Brief tone to notify the user that
  • Cisco SPA1001 | Administration Guide - Page 105
    Cadence script for distinctive CWT 3. The default is 30(.1/.1, .1/.1, .1/9.5). Cadence script for distinctive CWT 4. The default is 30(.1/.1, .3/.1, .1/9.3). Cadence script for distinctive CWT 5. The default is 30(.3/.1,.1/.1,.3/9.1). Document Version 3.2 Linksys IP Phone Administrator Guide 5-21
  • Cisco SPA1001 | Administration Guide - Page 106
    in an INVITE's Alert-Info Header to pick distinctive ring/CWT 7 for the inbound call. The default is Bellcore-r7. Name in an INVITE's Alert-Info Header to pick distinctive ring/CWT 8 for the inbound call. The default is Bellcore-r8. 5-22 Linksys IP Phone Administrator Guide Document Version 3.2
  • Cisco SPA1001 | Administration Guide - Page 107
    after receiving the first SIP 18x response before declaring the remote end is ringing. If a busy response is received during this time, the Linksys IP phone still considers the call as failed and keeps on retrying. The default is 0.5. Document Version 3.2 Linksys IP Phone Administrator Guide 5-23
  • Cisco SPA1001 | Administration Guide - Page 108
    This code calls the last caller. The default is *69. Redials the last number called. (Not in PAP2T) The default is *07. Begins a blind transfer of the current call to the extension specified after the activation code. The default is *98. 5-24 Linksys IP Phone Administrator Guide Document Version
  • Cisco SPA1001 | Administration Guide - Page 109
    the call ring through when do not disturb or call forwarding of all calls are enabled. The default is *64. Cancels the code to accept the last outbound call. The default is *84. Enables call waiting on all calls. The default is *56. Document Version 3.2 Linksys IP Phone Administrator Guide 5-25
  • Cisco SPA1001 | Administration Guide - Page 110
    Code Disables call waiting, caller ID generation. The default is *45. Dist Ring Act Code Enables the distinctive ringing feature. The default is *26 Dist Ring Deact Code Disables the distinctive ringing feature. The default is *46. 5-26 Linksys IP Phone Administrator Guide Document Version
  • Cisco SPA1001 | Administration Guide - Page 111
    this code. Toggles the line to a fax machine. (Not in PAP2T) The default is #99. These codes tell the Linksys IP phone what to do when the user places the current call on hold and is listening to the second dial tone. One or more *code can be configured into this parameter, such as *98, or *97
  • Cisco SPA1001 | Administration Guide - Page 112
    default. If the *code is not to be followed by a phone number, such as *73 to cancel call forwarding, do not include it in this parameter. In that case, simple add that *code in the dial plan and the Linksys IP phone send INVITE *73@..... as usual when user dials *73. Vertical Service Announcement
  • Cisco SPA1001 | Administration Guide - Page 113
    for the associated call. The default is *0272640. Makes this codec the preferred codec for the associated call. The default is *01729. Makes this codec the only codec that can be used for the associated call. The default is *02729. Document Version 3.2 Linksys IP Phone Administrator Guide 5-29
  • Cisco SPA1001 | Administration Guide - Page 114
    00, GMT+02:00, ..., GMT+13:00. The default is GMT-08:00. Sets the electrical impedance of the FXS port. Choices are 600, 900, 600+2.16uF, 900+2. seconds (0-59). The default Daylight Saving Time Rule is start=4/1/7;end=10/-1/7;save=1. 5-30 Linksys IP Phone Administrator Guide Document Version 3.2
  • Cisco SPA1001 | Administration Guide - Page 115
    PR (UK)-CID, CIDCW, and VMWI. FSK is sent after polarity reversal and DTAS and before first ring. Waits for ACK from CPE after DTAS for CIDCW. Polarity reversal is applied only if equipment is on hook. The default is Bellcore(N.Amer, China). The Linksys IP phone supports bell 202 and v.23 standards
  • Cisco SPA1001 | Administration Guide - Page 116
    5 Linksys ATA Field Reference Select the method you want to use, Default or Sweden default. (Not in PAP2T) The default is Default. Enable or disable more echo suppresion. The default is no. To use this test, select yes. Otherwise, keep the default, no. 5-32 Linksys IP Phone Administrator Guide
  • Cisco SPA1001 | Administration Guide - Page 117
    35 • Call Feature Settings, page 5-38 • Proxy and Registration, page 5-38 • Subscriber Information, page 5-40 • Supplementary Service Subscription, page 5-40 • Audio Configuration, page 5-42 • VoIP Fallback to PSTN (SPA3102/AG310), page 5-46 • Gateway Accounts (SPA3102/AG310), page 5-45 • Dial Plan
  • Cisco SPA1001 | Administration Guide - Page 118
    message, the Linksys IP phone ends this call with a SIP BYE message. The range is 0 to 255 seconds (0 means that the session refresh is disabled). The default is 30 mapped IP addresses and SIP/RTP ports in SIP messages, select yes. Otherwise, select no. The default is no. To send the configured NAT
  • Cisco SPA1001 | Administration Guide - Page 119
    . (Not in PAP2T) The default is 6. If enabled, the SPA will automatically route all calls to the PSTN gateway when the Line 1 proxy is down (registration failure or network link down). The default is yes. Determines how jitter buffer size is adjusted by the Linksys IP phone. Jitter buffer size is
  • Cisco SPA1001 | Administration Guide - Page 120
    IP phone 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. 5-36 Linksys IP
  • Cisco SPA1001 | Administration Guide - Page 121
    IP address. A source IP address is untrusted if it does not match any of the IP addresses resolved from the configured (or if is yes). The default is no. Controls when the Linksys IP phone sends BYE to terminate stale call legs upon completion of call
  • Cisco SPA1001 | Administration Guide - Page 122
    only a user ID is specified, the current or outbound proxy is contacted. Music-on-hold is disabled if the MOH Server is not specified. Makes the Linksys IP phone perform a transfer when a conference call has ended. Select yes or no from the drop-down menu. The default is yes. This feature supports
  • Cisco SPA1001 | Administration Guide - Page 123
    normal mode, this list will contain proxies ranked by weight and priority. if Based on SRV port is configured the PAP2T does normal first, and also inspect the port number based on 1st proxy's port on the list. The default is Normal. Document Version 3.2 Linksys IP Phone Administrator Guide 5-39
  • Cisco SPA1001 | Administration Guide - Page 124
    Line Tab Field Voice Mail Server Mailbox Subscribe Expires Description Enter the URL or IP address of the server. Chapter 5 Linksys ATA Field Reference Subscriber Information Field Display Name User ID Password Use Auth ID Auth ID Directory Number Call Capacity Mini Certificate
  • Cisco SPA1001 | Administration Guide - Page 125
    Disturb Service The default is yes. Enable Caller ID Service The default is yes. Enable Call Waiting Caller ID Service The default is yes. Enable Call Return Service The default is yes. Enable Call Redial Service (Not in PAP2T) Line Tab Document Version 3.2 Linksys IP Phone Administrator Guide
  • Cisco SPA1001 | Administration Guide - Page 126
    ) Call Transfer Service. The default is yes. Enable MWI Service. MWI is available only if a Voice Mail Service is set-up in the deployment. The default is yes. Enable VMWI Service (FSK). The default is yes. Enable Speed Dial Service. The default is yes. Enable Secure Call Service. The default is
  • Cisco SPA1001 | Administration Guide - Page 127
    no. The default is yes. To enable the use of the G.726 codec at 32 kbps, select yes. Otherwise, select no. The default is yes. To enable detection of the fax Calling Tone (CNG), select yes. Otherwise, select no. The default is yes. Document Version 3.2 Linksys IP Phone Administrator Guide 5-43
  • Cisco SPA1001 | Administration Guide - Page 128
    . The SPA8000 supports one T.38 connection for each of its four modules (Line 1-2, 3-4, 5-6, and 7-8). The SPA8000 supports a maximum of four connections, but it does not support two fax devices connected to the same module. The default is yes. 5-44 Linksys IP Phone Administrator Guide Document
  • Cisco SPA1001 | Administration Guide - Page 129
    default is no. This is the authentication user-id to be used by the SPA to authenticate itself to Gateway 1. The default is blank. This is the password to be used by the SPA to authenticate itself to Gateway 1. The default is blank. Document Version 3.2 Linksys IP Phone Administrator Guide 5-45
  • Cisco SPA1001 | Administration Guide - Page 130
    . If the user-id portion matches a pattern in the dial plan, then it is interpreted as a regular phone number according to the dial plan. The INVITE message, however, is still sent to the outbound proxy if it is enabled. Description 5-46 Linksys IP Phone Administrator Guide Document Version 3.2
  • Cisco SPA1001 | Administration Guide - Page 131
    connected: Forward or Reverse. The default is Forward. Polarity after an outbound call is connected: Forward or Reverse. The default is Forward. Polarity after an inbound call is connected: Forward or Reverse. The default is Forward. Document Version 3.2 Linksys IP Phone Administrator Guide 5-47
  • Cisco SPA1001 | Administration Guide - Page 132
    Line Tab Chapter 5 Linksys ATA Field Reference 5-48 Linksys IP Phone Administrator Guide Document Version 3.2
  • Cisco SPA1001 | Administration Guide - Page 133
    (SPA3102/AG310), page 5-53 • Subscriber Information (SPA3102/AG310), page 5-54 • Audio Configuration (SPA3102/AG310), page 5-55 • Dial Plans, page 5-57 • VoIP-To-PSTN Gateway Setup, page 5-57 • VoIP Users and Passwords (HTTP Authentication), page 5-59 • FXO (PSTN) Timer Values (sec), page 5-60
  • Cisco SPA1001 | Administration Guide - Page 134
    . Select the appropriate setting: up and down, up only, down only, or disable. The default is up and down. SIP Settings Field SIP Port Description Port number of the SIP message listening and transmission port. The default is 5060. 5-50 Linksys IP Phone Administrator Guide Document Version 3.2
  • Cisco SPA1001 | Administration Guide - Page 135
    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. Document Version 3.2 Linksys IP Phone Administrator Guide 5-51
  • Cisco SPA1001 | Administration Guide - Page 136
    IP address. A source IP address is untrusted if it does not match any of the IP addresses resolved from the configured (or if is yes). The default is no. Controls when the Linksys IP phone sends BYE to terminate stale call legs upon completion of call
  • Cisco SPA1001 | Administration Guide - Page 137
    to use DNS SRV lookup for Proxy and Outbound Proxy. The default is no. 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. Document Version 3.2 Linksys IP Phone Administrator Guide 5-53
  • Cisco SPA1001 | Administration Guide - Page 138
    yes. Otherwise, select no to use the user ID and password. The default is no. Authentication ID for SIP authentication. Maximum number of calls allowed on this line interface. Choices: {unlimited,1,2,3,...25 }. Default is 16. Note that the Linksys IP phone does not distinguish between incoming and
  • Cisco SPA1001 | Administration Guide - Page 139
    , select no. The default is yes. To enable the use of the echo suppressor, select yes. Otherwise, select no. The default is yes. To enable the use of the G726 codec at 24 kbps, select yes. Otherwise, select no. The default is yes. Document Version 3.2 Linksys IP Phone Administrator Guide 5-55
  • Cisco SPA1001 | Administration Guide - Page 140
    from the Hook Flash MIME Type setting. The default is None. If enabled, this feature automatically disables the echo canceller when a fax tone is detected. To use this feature, select yes. Otherwise, select no. The default is no. 5-56 Linksys IP Phone Administrator Guide Document Version 3.2
  • Cisco SPA1001 | Administration Guide - Page 141
    disable VoIP-To-PSTN Gateway functionality. The default is yes. 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 none. Document Version 3.2 Linksys IP Phone Administrator Guide 5-57
  • Cisco SPA1001 | Administration Guide - Page 142
    Note Authentication is skipped for Line 1 VoIP caller. The default is 1. Index of the dial plan in the dial plan pool to be used when the VoIP Caller is not authenticated. Choose from {none, 1, 2, 3, 4, 5, 6, 7, 8}. The default is 1. 5-58 Linksys IP Phone Administrator Guide Document Version 3.2
  • Cisco SPA1001 | Administration Guide - Page 143
    match any of the VoIP User Auth ID's, the INVITE will be rejected with a 403 response. The default is blank. 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. Document Version 3.2 Linksys IP Phone Administrator Guide 5-59
  • Cisco SPA1001 | Administration Guide - Page 144
    Field Default Ring Description 1-8, Follow Line Cfg FXO (PSTN) Timer Values (sec) Field VoIP Answer Delay Description Delay in seconds before auto-answering inbound VoIP calls for the FXO account. The range is 0-255. The default is 3. 5-60 Linksys IP Phone Administrator Guide Document
  • Cisco SPA1001 | Administration Guide - Page 145
    . The range is 0-2147483647. The default is 0. Limit on the duration of a VoIP-To-PSTN Gateway Call. Unit is in seconds. 0 means unlimited. The range is 0-2147483647. The default is 0. Delay after hook before the SPA dials a PSTN number. The range is 0-255. The default is 1. Delay after a ring burst
  • Cisco SPA1001 | Administration Guide - Page 146
    enabled, SPA will disconnect both call legs when it detects the disconnect tone from the PSTN side during a gateway call. Disconnect tone is specified in the parameter, which depends on the region of the PSTN service. The default is yes. 5-62 Linksys IP Phone Administrator Guide
  • Cisco SPA1001 | Administration Guide - Page 147
    limit of the ring frequency used to detect the ring signal. The default is 100. 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. The default is 0. Document Version 3.2 Linksys IP Phone Administrator Guide 5-63
  • Cisco SPA1001 | Administration Guide - Page 148
    default is high. 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. 5-64 Linksys IP Phone Administrator Guide Document
  • Cisco SPA1001 | Administration Guide - Page 149
    • @gw0 - forward to caller to the PSTN number (dialed automatically by the SPA through the PSTN gateway) The default is blank. Forward number for Call Forward Busy Service. Same as Cfwd All Dest. The default is blank. Document Version 3.2 Linksys IP Phone Administrator Guide 5-65
  • Cisco SPA1001 | Administration Guide - Page 150
    Caller number that is actively forwarded to by using the Call Forward Last activation code The default is blank. Cfwd Last Dest Forward number for the Same as Cfwd All Dest. The default is blank. 5-66 Linksys IP Phone Administrator Guide Document Version 3.2
  • Cisco SPA1001 | Administration Guide - Page 151
    user can also manually modify it to clear or set the flag. Setting this value to yes can activate stutter tone and VMWI signal. This parameter is stored in long term memory and will survive after reboot or power cycle. The default is no. Document Version 3.2 Linksys IP Phone Administrator Guide
  • Cisco SPA1001 | Administration Guide - Page 152
    hook. The default is None. Ring pattern for call back notification. The default is None. Duration of ring splash when a call is forwarded (0 - 10.0s). The default is 0. Duration of ring splash when a call is blocked (0 - 10.0s). The default is 0. 5-68 Linksys IP Phone Administrator Guide Document
  • Cisco SPA1001 | Administration Guide - Page 153
    splash when the VM server sends SIP NOTIFY message to the Linksys IP phone indicating that there are no more unread voicemails. Some equipment requires a short ring to precede the FSK signal to turn off VMWI lamp. The default is no. Document Version 3.2 Linksys IP Phone Administrator Guide 5-69
  • Cisco SPA1001 | Administration Guide - Page 154
    Linksys ATA Field Reference PSTN User Tab (SPA3102/AG310) This section describes the fields for the following headings on the PSTN User tab on the SPA3102: • PSTN-To-VoIP Selective Call Forward Settings, page 5-70 • PSTN-To-VoIP Speed Dial Linksys IP Phone Administrator Guide Document Version 3.2
  • Cisco SPA1001 | Administration Guide - Page 155
    by Line 1's distinctive ring settings. The default is 1. PSTN/VoIP Caller Commands via DTMF Field **# **1 Description Disconnect the PSTN line (SPA will take the FXO port on-hook). End the current call and restarts dial tone. Document Version 3.2 Linksys IP Phone Administrator Guide 5-71
  • Cisco SPA1001 | Administration Guide - Page 156
    PSTN User Tab (SPA3102/AG310) Chapter 5 Linksys ATA Field Reference 5-72 Linksys IP Phone Administrator Guide Document Version 3.2
  • Cisco SPA1001 | Administration Guide - Page 157
    CAS CDR CID CIDCW CNG CPC CPE CWCID CWT D/A dB dBm DHCP DNS DRAM DSL DSP DTAS DTMF FQDN FSK Auto-Configuration Server Analog To Digital Converter Anonymous Call Back to Back User Agent Boolean Values. Specified as "yes" and "no", or "1" and "0" in the profile Certificate Authority CPE Alert Signal
  • Cisco SPA1001 | Administration Guide - Page 158
    FXS GW ITU HTML HTTP HTTPS ICMP IGMP ILEC IP ISP ITSP IVR LAN LBR LBRC MC MGCP MOH MOS ms Management Protocol Incumbent Local Exchange Carrier Internet Protocol Internet Service Provider IP Telephony Service Provider Interactive Voice Public Switched Telephone Network Network Address Translation Out-
  • Cisco SPA1001 | Administration Guide - Page 159
    SDP SDRAM sec SIP SLA SLIC SP SPA SSL TFTP TCP UA uC UDP URL VM VMWI VQ WAN XML Session Description Protocol Synchronous DRAM seconds Session Initiation Protocol Shared line appearance Subscriber Line Interface Circuit Service Provider Linksys Phone Adaptor Secure Socket Layer Trivial File Transfer
  • Cisco SPA1001 | Administration Guide - Page 160
    Appendix A Acronyms Linksys ATA Administrator Guide A-4 Document Version 3.2
  • Cisco SPA1001 | Administration Guide - Page 161
    of equipment are telephones, key systems, PBXs, voice-mail systems, and call accounting systems as well as wiring telephone jacks. The standard for this equipment is set by the FCC, and the equipment is supplied by an interconnect company. Document Version 3.01 Linksys ATA Administrator Guide B-1
  • Cisco SPA1001 | Administration Guide - Page 162
    - Platform that lets users send, receive, and manage all e-mail, voice, and fax messages from any telephone, PC, or information device. Voicemail-A system that allows storage and retrieval of voice messages through voice-mail boxes. Linksys ATA Administrator Guide B-2 Document Version 3.01
  • Cisco SPA1001 | Administration Guide - Page 163
    plan and matches it against the cumulative number entered by the user. The SPA also detects invalid phone numbers not compatible with the dial plan and alerts the user via a configurable tone (Reorder) or announcement. Hang up the telephone. Document Version 3.01 Linksys ATA Administrator Guide
  • Cisco SPA1001 | Administration Guide - Page 164
    164 ID (phone number) so that they may be reached from wired or wireless callers on the PSTN or IP network. The SPA supplies ring voltage to the attached telephone set to alert the user of incoming calls. Hang up the telephone. Enhanced Services Caller ID Service description User action required
  • Cisco SPA1001 | Administration Guide - Page 165
    3. Press *81 4. Listen for dial tone 5. Dial the telephone number you are calling You must repeat this process at the start of each call. Expected call and network behavior The user activates this service to hide their Caller ID when making an outgoing call. User action required to No action
  • Cisco SPA1001 | Administration Guide - Page 166
    user action is required. If you deactivated call waiting and wish to reinstate the service, do the following: 1. Lift the receiver 2. Listen for dial tone 3. Press *57 You will hear a confirmation tone signaling your request to cancel Call Waiting has been accepted. Linksys ATA Administrator Guide
  • Cisco SPA1001 | Administration Guide - Page 167
    prompted by a voice response system with instructions to listen to your messages. Expected call and network behavior When voice mail is available for a subscriber, a notification message is sent from the voice mail server to the SPA. When the user dials their own phone number, the SPA connects the
  • Cisco SPA1001 | Administration Guide - Page 168
    pressing the switch hook or flash button, creating a three-way conference. When the user hangs up the phone, the transferee and the called party remain in a call. Not applicable. Unattended or "Blind" Call Transfer Service description User action required to activate or use Unattended or "Blind
  • Cisco SPA1001 | Administration Guide - Page 169
    or flash button. Expected call and network behavior The SPA supports up to two calls per line. The SPA can conference two calls by bridging the second and third parties. User action required to Hang up the telephone. deactivate or end Document Version 3.01 Linksys ATA Administrator Guide C-7
  • Cisco SPA1001 | Administration Guide - Page 170
    the original caller to the third party Call Return Service description User action required to activate or use Expected call and network behavior User action required to deactivate or end The SPA supports a service that allows the SPA to automatically dial the last caller's number. 1. Pick up the
  • Cisco SPA1001 | Administration Guide - Page 171
    deactivate or end 2. Listen for dial tone 3. Press *__ Call FWD-Unconditional Service description All calls are immediately forwarded to the designated forwarding number. The SPA does not ring or provide call waiting when Call FWD-Unconditional is activated. User action required to 1. Lift the
  • Cisco SPA1001 | Administration Guide - Page 172
    This feature allows a user the option to divert (forward) calls to their telephone number to any number when their phone is busy or in conference by using the touchtone keypad of their telephone or web browser interface. This service is activated or deactivated from the phone being forwarded or the
  • Cisco SPA1001 | Administration Guide - Page 173
    C User Guidelines Call FWD-No Answer Enhanced Services Service description User action required to activate or use Calls are forwarded to the designated forwarding number after a configurable time period elapses while the SPA is ringing and does not answer. 1. Lift the receiver 2. Listen for dial
  • Cisco SPA1001 | Administration Guide - Page 174
    tone (that is, short-long-short) accompanies the incoming call. Calls from other telephone numbers ring normally. Speed Calling-Up to Eight Numbers or IP Addresses Service description User action required to activate or use The SPA supports user programming of up to eight long distance, local
  • Cisco SPA1001 | Administration Guide - Page 175
    Appendix C User Guidelines Enhanced Services Expected call and network 1. Pick up the receiver behavior 2. Listen for dial tone 3. Press single digit code assigned to the stored number (2-9) 4. Press # to signal dialing complete The number is automatically dialed normally. User action
  • Cisco SPA1001 | Administration Guide - Page 176
    Enhanced Services Appendix C User Guidelines C-14 Linksys ATA Administrator Guide Document Version 3.01
  • Cisco SPA1001 | Administration Guide - Page 177
    accessing IVR 2-18 accounts, User and Administrator 2-23 administration budget 2-14 binary format 3-4 Blind Attn-Xfer Enable 5-38 Blind Transfer Code 5-24 Block ANC Act Code 5-26 5-26 Block CID Per Call Act Code 5-26 Block CID Per Call Deact Code 5-26 Block Linksys ATA Administrator Guide IN-1
  • Cisco SPA1001 | Administration Guide - Page 178
    Call 1 Duration 5-4 Call 1 Encoder 5-4 Call 1 FAX 5-4 Call 1 Jitter 5-4 Call 1 Packet Error 5-4 Call 1 Packets Lost 5-4 Call 1 Packets Recv 5-4 Call 1 Packets Sent 5-4 Call 1 Peer Name 5-4 Call 1 Peer Phone 5-4 Call 1 Remote Hold 5-4 Call 1 Round Trip Delay 5-4 Call 1 State 5-4 Call 1 Tone 5-4 Call
  • Cisco SPA1001 | Administration Guide - Page 179
    2-19 primary DNS server 2-20 static gateway IP address 2-20 WAN IP address 2-19 CID_Serv parameter 5-41 CID Act Code 5-26 CID Deact Code 5-26 CID Setting parameter 5-67 Conference Act Code 5-27 Conference Tone 5-20 configuration saving 3-19 configuration profile 3-4 Confirm Tone 5-20 CPC Delay 5-24
  • Cisco SPA1001 | Administration Guide - Page 180
    Ethernet port check 2-20 examples dial plans 3-8 External IP parameter 5-3 EXT IP 5-17 EXT RTP Port Min 5-18 IN-4 Linksys ATA Administrator Guide EXT SIP Port parameter 5-36, 5-51 F Factory 2-21 factory defaults resetting 2-21 user 2-21 FAX CED Detect Enable 5-43, 5-56 FAX CNG Detect Enable 5-43
  • Cisco SPA1001 | Administration Guide - Page 181
    , in dial plans 3-8 INFOREQ Dynamic Payload 5-15 Insert VIA received 5-17 Insert VIA rport 5-17 Interactive Voice Response see depend on configuration 2-18 options 2-19 password 2-18 quick reference 2-18 time out 2-19 J Jitter Buffer Adjustment 5-35, 5-50 Linksys ATA Administrator Guide IN-5
  • Cisco SPA1001 | Administration Guide - Page 182
    Waiting 5-4 Message Waiting parameter 5-67 Mini Certificate parameter 5-40, 5-54 Modem Line Toggle Code 5-27 IN-6 Linksys ATA Administrator Guide MOH Server 5-38 multicast address check 2-20 set 2-20 MWI Dial Tone 5-20 MWI Serv parameter 5-42 N NAT Keep Alive Dest 5-35, 5-50 NAT Keep Alive Enable
  • Cisco SPA1001 | Administration Guide - Page 183
    63 PSTN-To-VoIP Call Max Dur parameter 5-61 Q quick-reference for IVR 2-18 R Real-Time Protocol 1-10 reboot 2-20 Referee Bye Delay 5-37, 5-52 Referor Bye Delay 5-37, 5-52 Referral Services Codes 5-27 Referral Serv parameter 5-42 Refer Target Bye Delay 5-37, 5-52 Refer-To Target Contact 5-38, 5-53
  • Cisco SPA1001 | Administration Guide - Page 184
    Index Remove Last Reg 5-11 Reorder Delay 5-23 Reorder Tone 5-19 repetition, in dial plans 3-7 resetting factory defaults 2-21 Restricted Access Domains parameter 5-8 Restrict Source IP 5-37, 5-52 Resync URL 3-3 Retry Reg RSC 5-14 Ring1/2/3/4/5/6/7/8 Caller parameter 5-68, 5-70 Ring1 Cadence 5-21
  • Cisco SPA1001 | Administration Guide - Page 185
    Key parameter 5-40, 5-54 static gateway IP address check 2-20 static IP addressing check 2-19 Static IP parameter 5-8 Stats In BYE 5-15 Sticky 183 parameter 5-38, 5-53 STUN Enable 5-17 STUN Server 5-17 STUN Test Enable 5-17 sub-sequence in dial plans 3-7 Substitute VIA Addr 5-17 Symmetric RTP
  • Cisco SPA1001 | Administration Guide - Page 186
    user factory defaults 2-21 User ID 5-40, 5-54 User ID parameter 5-3 User Password 5-8 V VMWI Refresh Intvl 5-24 VMWI Ring Policy parameter 5-69 VMWI Ring Splash Len parameter 5-69 VMWI Serv parameter 5-42 voice codecs 2-14 VoIP Access List parameter 5-59 VoIP Answer Delay parameter 5-60 VoIP Call
  • 1
  • 2
  • 3
  • 4
  • 5
  • 6
  • 7
  • 8
  • 9
  • 10
  • 11
  • 12
  • 13
  • 14
  • 15
  • 16
  • 17
  • 18
  • 19
  • 20
  • 21
  • 22
  • 23
  • 24
  • 25
  • 26
  • 27
  • 28
  • 29
  • 30
  • 31
  • 32
  • 33
  • 34
  • 35
  • 36
  • 37
  • 38
  • 39
  • 40
  • 41
  • 42
  • 43
  • 44
  • 45
  • 46
  • 47
  • 48
  • 49
  • 50
  • 51
  • 52
  • 53
  • 54
  • 55
  • 56
  • 57
  • 58
  • 59
  • 60
  • 61
  • 62
  • 63
  • 64
  • 65
  • 66
  • 67
  • 68
  • 69
  • 70
  • 71
  • 72
  • 73
  • 74
  • 75
  • 76
  • 77
  • 78
  • 79
  • 80
  • 81
  • 82
  • 83
  • 84
  • 85
  • 86
  • 87
  • 88
  • 89
  • 90
  • 91
  • 92
  • 93
  • 94
  • 95
  • 96
  • 97
  • 98
  • 99
  • 100
  • 101
  • 102
  • 103
  • 104
  • 105
  • 106
  • 107
  • 108
  • 109
  • 110
  • 111
  • 112
  • 113
  • 114
  • 115
  • 116
  • 117
  • 118
  • 119
  • 120
  • 121
  • 122
  • 123
  • 124
  • 125
  • 126
  • 127
  • 128
  • 129
  • 130
  • 131
  • 132
  • 133
  • 134
  • 135
  • 136
  • 137
  • 138
  • 139
  • 140
  • 141
  • 142
  • 143
  • 144
  • 145
  • 146
  • 147
  • 148
  • 149
  • 150
  • 151
  • 152
  • 153
  • 154
  • 155
  • 156
  • 157
  • 158
  • 159
  • 160
  • 161
  • 162
  • 163
  • 164
  • 165
  • 166
  • 167
  • 168
  • 169
  • 170
  • 171
  • 172
  • 173
  • 174
  • 175
  • 176
  • 177
  • 178
  • 179
  • 180
  • 181
  • 182
  • 183
  • 184
  • 185
  • 186
Corporate Headquarters
Linksys
121 Theory Drive
Irvine, CA 92617
USA
Tel: 949 823-1200
800 546-5797
Fax: 949 823-1100
Linksys ATA
Administrator Users Guide
Document Version 3.2