Cisco SPA2102-SF Administration Guide

Cisco SPA2102-SF Manual

Cisco SPA2102-SF manual content summary:

  • Cisco SPA2102-SF | Administration Guide - Page 1
    ADMINISTRATION GUIDE Cisco Small Business SPA2102, SPA3102, SPA8000, SPA8800, PAP2T Analog Telephone Adapters
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    , Inc. and/or its affiliates in the U.S. and other countries. A listing of Cisco's trademarks can be found at www.cisco.com/go/trademarks. Third party trademarks mentioned are the property of their respective owners. The use of the word partner does not imply a partnership relationship
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    Basic Services and Equipment Required 36 Downloading Firmware 37 Basic Installation and Configuration 37 Upgrading the Firmware for the ATA Device 37 Setting up Your ATA Device 38 Using the Administration Web Server 39 Connecting to the Administration Web Server 40 Setting Up the WAN
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    IP Address 48 Configuring NAT Mapping with STUN 50 Determining the Router's NAT Mechanism 52 Firewalls and SIP 53 Configuring SIP Timer Values 53 Chapter 4: Configuring Voice Services 54 Supported Codecs 54 Using a FAX Machine 55 Fax Troubleshooting 56 Managing Caller ID Service
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    Audio Server 88 About the Streaming Audio Server 88 Configuring the Streaming Audio Server 90 Using the IVR with an SAS Line 91 Chapter 6: Configuring the PSTN (FXO) Gateway on the SPA3102 92 Connecting to PSTN and VoIP Services 92 How VoIP-To-PSTN Calls Work 93 One-Stage Dialing
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    DHCP Lease Settings section Application page Port Forwarding Settings section DMZ Settings section Miscellaneous Settings section System Reserved Ports Range section Appendix B: ATA Voice Field Reference Info page Product Information section System Status section Line Status section Cisco Small
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    Tone Spec section Control Timer Values (sec) section Vertical Service Activation Codes section Vertical Service Announcement Codes section (SPA2102, SPA8000) Outbound Call Codec Selection Codes section Miscellaneous section Line page Line Enable section Streaming Audio Server (SAS) section NAT
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    section (SPA8800) Call Forward, Speed Dial, Supplementary Services, and Ring Settings (SPA8000 and SPA8800) Trunk Group page (SPA8000) Line Enable section Network Settings section SIP Settings section Subscriber Information section Dial Plan section NAT Settings section Proxy and Registration
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    Audio Configuration section Dial Plans section VoIP-To-PSTN Gateway Setup section VoIP Users and Passwords (HTTP Authentication) section Ring Settings section FXO (PSTN) Timer Values (sec) section PSTN Disconnect Detection section International Control (Settings) section User page Call Forward
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    Product Resources Related Documentation Contents 237 238 Cisco Small Business ATA Administration Guide 10
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    System administrators or anyone who performs LVS installation and administration NOTE This guide does not provide the configuration information required by specific service providers. Please consult with the service provider for specific service parameters. Cisco Small Business ATA Administration
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    on • A literal value to be entered in a field. May indicate either of the following: • A variable that should be replaced with a literal value. • The name of a page, section, or field in the user interface Indicates code samples or system output. Cisco Small Business ATA Administration Guide 12
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    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
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    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 , and choose the options you want. d. Click Search. Cisco Small Business ATA Administration Guide 14
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    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. Cisco Small Business ATA Administration Guide 15
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    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
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    • "ATA Connectivity Requirements," on page 20 • "ATA Software Features," on page 26 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
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    . Wireless-G IP router with two FXS ports. Provides ATA device functionality. 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
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    Introducing Cisco Small Business Analog Telephone Adapters Comparison of ATA Devices Figure 3 How ATAs Provide Voice Connectivity Ethernet/Wireless LAN 1 Fax (up to 4 SPA8000) W WRTP54G and SPA2102 PSTN Broadband router DSL/cable modem WSPA3102 Internet Analog phone (up to 8 with SPA8000)
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    the following functionality to be supplied on the network connected to its Ethernet port: • Connection to an IP router with hairpinning support • Connection to an outbound Proxy server When a phone connected to the ATA device communicates with another phone, it sends a SIP packet onto the internal
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    ISP LAN WAN Internet IP ITSP NOTE • The IVR functions are accessed by connecting an analog telephone to Line 1. • For proper operation, the service provider should use an Outbound Proxy to forward all voice traffic when the PAP2T is located behind a router. If necessary, explicit port ranges can
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    The IVR functions are accessed by connecting an analog telephone to Line 1. • For proper operation, the service provider should use an Outbound Proxy to forward all voice traffic when the SPA2102 is located behind a router. If necessary, explicit port ranges can be specified for SIP and RTP. Cisco
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    The IVR functions are accessed by connecting an analog telephone to Line 1. • For proper operation, the service provider should use an Outbound Proxy to forward all voice traffic when the SPA3102 is located behind a router. If necessary, explicit port ranges can be specified for SIP and RTP. Cisco
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    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 pairs of RJ-11 ports. Up to four fax machines can be connected to the SPA8000 router, but they must be distributed as shown. Cisco Small Business ATA Administration Guide
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    Cisco Small Business Analog Telephone Adapters ATA Connectivity Requirements 1 NOTE • You can use line 1 or line 2 to access the IVR functions. • For proper operation, the service provider should use an Outbound Proxy to forward all voice traffic when the SPA8000 is located behind a router
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    forward all voice traffic when the SPA8800 is located behind a router. If necessary, explicit port ranges can be specified for SIP and RTP. ATA Software Features The ATA device is a full featured, fully programmable phone adapter that can be custom provisioned within a wide range of configuration
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    at one through ten 10 ms voice frames per packet. This codec provides high voice quality. G.729a The ITU G.729 voice coding algorithm is used to compress digitized speech. Cisco supports G.729. G.729a is a reduced complexity version of G.729. It requires about half the processing power to code
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    . Feature Streaming Audio Server T.38 Fax Relay Silence Suppression Description See "Configuring a Streaming Audio Server," on page 88. See "Using a FAX Machine," on page 55. See "Silence Suppression and Comfort Noise Generation," on page 60. Cisco Small Business ATA Administration Guide 28
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    more gradually. If jitter level is lower, it shrinks more quickly. Adaptive Jitter Buffer is configured in the Line and PSTN Line tabs. See "ATA Voice Field Reference," on page 115. International Caller ID Delivery In addition to support of the Bellcore (FSK) and Swedish/ Danish (DTMF) methods of
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    feature allows the user to hear the call progress tones (such as ringing) that are generated from the far-end network. See the Regional tab in the "ATA Voice Field Reference," on page 115. Echo Cancellation Impedance mismatch between the telephone and the IP Telephony gateway phone port can lead
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    Service (parameter three-way call serv set in the Line tab) You can configure the length of time allowed for detection of a hook flash using the Hook Flash Timer parameter on the Regional tab of the administration web server. See "ATA Voice Field Reference," on page 115. Configurable Dial Plan
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    new ports do not need to be opened or packets dropped, because TCP is already in use for basic activities such as Internet browsing or e-commerce. SIP over TCP is configured in the Line tabs. See "ATA Voice Field Reference," on page 115. This feature applies to SPA2102, SPA3102, and SPA8800. Cisco
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    back to the transmitter/receiver instead of transmitting the data sampled on its local microphone (IP phone) or attached analog telephone (ATA-type device). Media loopback is configured in the User tab. See "ATA Voice Field Reference," on page 115. Cisco Small Business ATA Administration Guide 33
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    feature is enabled, the Reg Retry Random Delay setting is added on top of the exponential back-off adjusted delay value. The default value is 0, which disables the exponential back-off feature. Register Retry is configured in the SIP tab. See "ATA Voice Field Reference," on page 115. Cisco Small
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    Introducing Cisco Small Business Analog Telephone Adapters ATA Software Features 1 Feature DHCP Renewal on Timeout Description 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
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    page 36 • "Downloading Firmware" section on page 37 • "Basic Installation and Configuration" section on page 37 • "Upgrading the Firmware for the ATA Device" section on page 37 • "Setting up Your ATA Device" section on page 38 • "Using the Administration Web Server" section on page 39 • "Upgrading
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    : a. Connect an analog telephone to the Phone 1 or Phone 2 port on the ATA device. (You may not hear a dial tone. Continue to step b.) b. Press **** on the keypad to access the IVR menu. c. Press 110# to determine the Internet (WAN) IP address. Cisco Small Business ATA Administration Guide 37
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    the upgrade, point the web browser to the IP address of the ATA device. Check the Router > Status page. The Software Version field should show the firmware version that you installed. NOTE You may need to refresh your browser to display the updated page reflecting the new version number. Setting up
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    ATA device. It includes the following topics: • "Connecting to the Administration Web Server" section on page 40 • "Setting Up the WAN Configuration for Your ATA Device" section on page 40 • "Registering to the Service Provider" section on page 42 Cisco Small Business ATA Administration Guide 39
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    to the Phone 1 port of the ATA device. (You may not hear a dial tone. Continue to step b.) b. Press **** on the keypad to access the IVR menu. c. Press 110# to determine the Internet (WAN) IP address. NOTE For more information on the IVR menu, see your Quick Installation Guide or User Guide for your
  • Cisco SPA2102-SF | Administration Guide - Page 41
    Web Server 2 STEP 3 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
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    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
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    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. Cisco
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    syntax of the Resync URL is as follows: http://spa-ip-addr/admin/resync?[[protocol://][server-name[:port]]/profilepathname] NOTE The SPA resyncs only when it is idle. If no parameter follows /resync?, the Profile Rule setting from the Provisioning page is used. If no protocol is specified, TFTP is
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    or HTTP (firmware upgrades using HTTPS are not supported). Remote upgrades are controlled by configuring the desired firmware image URL into the ATA device via a remote profile resync. For further information about remote provisioning refer to the SPA Provisioning Guide. Cisco Small Business ATA
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    tool: Admin_Passwd "some secret"; Upgrade_Enable "Yes"; Refer to the SPA Provisioning Guide for further details. 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
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    that your infrastructure properly supports voice services. • "Network Address Translation (NAT) and Voice over IP (VoIP)," on page 47 • "Firewalls and SIP," on page 53 • "Configuring SIP Timer Values," on page 53 Network Address Translation (NAT) and Voice over IP (VoIP) NAT is a function that
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    on the ports to which the SPA devices are connected. NOTE Use NAT mapping only if the ITSP network does not provide a Session Border Controller functionality. STEP 1 Connect to the administration web server, and choose Admin access with Advanced settings. STEP 2 Click Voice tab > SIP. Cisco Small
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    Send Resp To Src Port: yes • EXT IP: Enter the public IP address for your router. Voice tab > SIP: NAT Support Parameters STEP 4 Click Voice tab > Line N, where N represents the line interface number. STEP 5 Scroll down to the NAT Settings section. • NAT Mapping Enable: Choose YES. • NAT Keep Alive
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    . STEP 1 Connect to the administration web server, and choose Admin access with Advanced settings. STEP 2 Click Voice tab > SIP. STEP 3 Scroll down to the NAT Support Parameters section, and then enter the following settings to enable and support the STUN server settings: • Handle VIA received: yes
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    Configuring Your System for ITSP Interoperability Network Address Translation (NAT) and Voice over IP (VoIP) Voice tab > SIP > NAT Support Parameters 3 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
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    syslog server is configured and is ready to receive syslog messages. STEP 1 Make sure you do not have firewall running on your PC that could block the syslog port (port 514 by default). STEP 2 Connect to the administration web server, and choose Admin access with Advanced settings. STEP 3 To enable
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    enter a longer SIP T1 retry value. To view the default settings or to make changes, open the Voice > SIP page, and scroll down to the SIP Timer Values section. For field descriptions, see "SIP Timer Values (sec) section," on page 129 of Appendix B. Cisco Small Business ATA Administration Guide 53
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    for voice services. • "Supported Codecs," on page 54 • "Using a FAX Machine," on page 55 • "Managing Caller ID Service," on page 58 • "Silence Suppression and Comfort Noise Generation," on page 60 • "Configuring Dial Plans," on page 61 • "Secure Call Implementation," on page 71 • "SIP Trunking
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    Configuring Voice Services Using a FAX Machine 4 • G.726-40 • G.729a • G.723 WRTP54G • G.711u (configured by default) • G.711a • G.726-32 • G.729a • G.723 Using a FAX Machine You can connect a fax machine to an FXS port on the SPA2102, SPA3102, SPA8000, and SPA8800. T.38 Fax is supported on these
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    6 If you are using a Cisco media gateway 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 Fax Troubleshooting If have problems sending or receiving
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    : • Jitter • Loss • Delay STEP 5 If faxes fail consistently, capture a copy of the web interface settings by selecting Save As > Web page, complete from the administration web server page. You can send this configuration file to Technical Support. STEP 6 Enable and capture the debug log. For
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    Configuring Voice Services Managing Caller ID Service 4 Managing Caller ID Service The choice of caller ID (CID) method is dependent on your area/region. To configure DTAS) and before first ring. The default is Bellcore(N.Amer, China). The ATA device supports bell 202 and v.23 standards for caller
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    Configuring Voice Services Managing Caller ID Service 4 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
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    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
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    of sequences is enclosed within parentheses. Each digit sequence within the dial plan consists of a series of elements, which are individually matched to the keys that the user presses. NOTE White space is ignored, but may be used for readability. Cisco Small Business ATA Administration Guide 61
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    Configuring Voice Services Configuring Dial Plans 4 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
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    : need example NOTE The SPA implicitly appends the vertical code sequences entered in the regional parameter settings to the end of the dial plan. Likewise, if Enable_IP_Dialing is enabled, then IP dialing is also accepted on the associated line. Cisco Small Business ATA Administration Guide 63
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    Configuring Voice Services Configuring Dial Plans 4 Digit Sequence Examples The following examples show digit sequences that you can enter in a dial plan. In a complete dial plan entry, sequences are separated by a pipe character (|), and the entire set of sequences is enclosed within parentheses.
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    Configuring Voice Services Configuring Dial Plans 4 8, xxxxxxx This is example is useful where a local area code is required by the carrier but the majority of calls go to one area code. After the user presses 8, an external dial tone sounds. The user can enter any seven-digit number. The
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    Voice Services Configuring Dial Plans 4 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
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    taking a phone off hook, a user has 9 seconds to begin dialing. If no digits are pressed within 9 seconds, the user hears a reorder (fast busy) tone. By setting a longer timer, you allow more time for users to enter the digits. • Create a hotline for all sequences on the System Dial Plan EXAMPLE
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    one digit sequence in the dial plan. Unless the user enters another digit within the specified number of seconds, the entry is evaluated. If it is valid, the call proceeds. If it is invalid, the call is rejected. The default value is 3 seconds. Cisco Small Business ATA Administration Guide 68
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    within the dial plan. EXAMPLE: (9,8[2-9]xxxxxx | 9,8,1[2-9]xxxxxxxxxS0 | 9,8,011xx. | 9,8,xx.|[1-8]xx) 9,8,1[2-9]xxxxxxxxxS0 With the timer set to 0, the call is transmitted automatically when the user dials the final digit in the sequence. Cisco Small Business ATA Administration Guide 69
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    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
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    Secure Call Indication Tone for a short time to indicate to both parties that the call is secured and that RTP traffic in both directions is being encrypted. If the user has a phone that supports call waiting caller ID (CIDCW) and that service is enabled, the CID will be updated with the information
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    Configuring Voice Services Secure Call Implementation 4 The ATA device can be configured so that, by default, all outbound calls are either secure or not secure. If secure by default, the user has the option to disable security when making a call by dialing *19 before dialing the target number. If
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    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
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    Configuring Voice Services Secure Call Implementation 4 NOTE The partner sites require a logon. The Mini Certificate Generator uses the following syntax: gen_mc ca-key user-name user-id expire-date Where: • ca-key is a text file be provisioned. EXAMPLE Cisco Small Business ATA Administration Guide
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    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
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    lines to the PBX. Fax PBX System Fax 194488 PBX System SPA8000 Integrated Access Device Internet ITSP The SPA8000 offers four trunk groups, numbered T1, T2, T3, and T4. A SIP-based voice service with an ITSP can be configured on each trunk group with a distinct phone number. Each of the
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    Configuring Voice Services SIP Trunking and Hunt Groups on the SPA8000 4 The following figure shows a simplified logical block diagram of the SPA8000 with the SIP Trunking feature. Figure 1 Logical Block Diagram of SIP Trunking Phone 1 L1 Phone 2 L2 Phone 3 L3 Phone 4 L4 Phone 5 L5 Phone 6 L6 ITSP
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    to ring (that is, for the proxy to send SIP INVITE to), and starts "hunting." (See "Configuring a Trunk Group," on page 82) STEP 3 When a line is selected to ring, one or more PBX phones may be alerted, according to the PBX features and configuration. STEP 4 The Caller ID of the external Caller is
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    to the particular trunk group for this Contact List. • You can instruct the SPA8000 to hunt only the phones that are on-hook, through the Voice tab > SIP page, Trunking Parameters section, Hunt Policy field. See "Setting the Hunt Policy," on page 85. • hunt=hrule: The hunt order, ring interval, and
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    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
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    Configuring Voice Services SIP Trunking and Hunt Groups on the SPA8000 4 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
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    the administration web server, and choose Admin access with Advanced settings. STEP 2 Assign each line to a trunk group, as needed: a. Click Voice tab > Ln, where n represents the number of the line interface. b. In the Trunk Group field, near the top of the line configuration page, choose a trunk
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    the top right corner of the web page and at the lower left corner. You also can connect directly to the Trunk Status Page by entering the following URL: http://spa8000-ip-addr/status. This page is available with the User Login or the Admin Login. Cisco Small Business ATA Administration Guide 83
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    Configuring Voice Services SIP Trunking and Hunt Groups on the SPA8000 Trunk Status page 4 The Trunk Status page shows all calls that are currently active on each trunk group. This page button on the web browser toolbar. The page shows the following Cisco Small Business ATA Administration Guide 84
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    Configuring Voice Services SIP Trunking and Hunt Groups on the SPA8000 4 Setting the Hunt Policy You can configure the SPA8000 so that the hunt rule applies to all phone or only to the phones that are on hook. STEP 1 Connect to the administration web server, and choose Admin access with Advanced
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    ," on page 86 • "Changing the Music File for the Internal Music Source," on page 87 Using the Internal Music Source To use the internal music source, simply identify imusic as the MOH server for each IP phone. STEP 1 Use the phone menu to find the IP address of the phone: a. Press the Setup button
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    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
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    topics: • "About the Streaming Audio Server," on page 88 • "Configuring the Streaming Audio Server," on page 90 • "Using the IVR with an SAS Line," on page 91 About the Streaming Audio Server The Streaming Audio Server (SAS) feature lets you attach an audio source to an FXS port and use it as
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    ." • If an incoming call is auto-answered, but later the FXS port changes to on-hook, the call is not terminated but continues to stream silence packets to the caller. • The SAS line can be set up to refresh each streaming audio session periodically using a SIP re-INVITE message, which detects if
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    as Receptionist Area Fax Machine. • User ID: Enter a three- to four-digit extension number that is not is use by another extension. c. In the Streaming Audio Server (SAS) section, choose yes from the SAS Enable drop-down list. STEP 4 Click Submit All Changes. STEP 5 Configure each phone to use this
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    Hold Configuring a Streaming Audio Server g. Close the window for the Telephone Configuration page. h. Repeat this step to configure each phone, as needed. 5 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
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    as a PSTN gateway and VoIP gateway, without relying on a PBX for call control. - FXS Port: The SPA3102 has 1 FXS port that you can connect to a standard analog telephone or fax machine. Configure the FXS settings by using the SPA3102 Line page. Cisco Small Business ATA Administration Guide 92
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    parameter must match one of the VoIP accounts stored on the ATA device. You can configure these settings on the Voice tab > PSTN Line page. For more information, see "VoIP Users and Passwords (HTTP Authentication) section," on page 212. Cisco Small Business ATA Administration Guide 93
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    tone right after the call is answered (by a SIP 200 response). You also can enable PIN authentication. In this case, the VoIP caller is prompted to enter a PIN number after the SPA3102 answers the call. The PIN number must end with a # key. The inter-PIN-digit timeout is 10 seconds (not configurable
  • Cisco SPA2102-SF | Administration Guide - Page 95
    number. (These dial plans are configured on the Voice Voice tab > PSTN Line page, Dial Plans section.) • If the PIN does not match one of the configured PSTN PIN values, then the ATA device plays the reorder tone and then takes the FXO port on-hook. Cisco Small Business ATA Administration Guide 95
  • Cisco SPA2102-SF | Administration Guide - Page 96
    Line page. For more information, see "PSTN Disconnect Detection section," on page 215. On the SPA8800, these settings can be configured on the Voice tab > Phone page. For more information, see "FXS Port Polarity Configuration section," on page 182. Cisco Small Business ATA Administration Guide 96
  • Cisco SPA2102-SF | Administration Guide - Page 97
    this behavior by IP dialing, through which the calls can be routed to any IP address that the user enters. The SPA3102 allows flexible call routing with four sets of gateway parameters and configurable dial plans. These settings are configured on the SPA3102 Voice tab > Line page. See "Gateway
  • Cisco SPA2102-SF | Administration Guide - Page 98
    below. For SPA3102, you can configure this setting on the Voice tab > Line page. For SPA8800, you can configure this setting on the Voice tab > Phone page. For more information, see "VoIP Fallback to PSTN section (SPA3102 and SPA8800)," on page 180. Sharing One VoIP Account Between the FXS and PSTN
  • Cisco SPA2102-SF | Administration Guide - Page 99
    VoIP gateway picks up the call. Symmetric RTP (SPA3102 and SPA8800) The Symmetric RTP parameter is used to send audio RTP to the source IP and port of the inbound RTP packets. This facilitates NAT traversal. See "SIP Parameters section," on page 127. Cisco Small Business ATA Administration Guide
  • Cisco SPA2102-SF | Administration Guide - Page 100
    When one VoIP account is shared between the FXS and PSTN Lines, the following parameters are recommended to be set. For more information, see the Regional page in the "ATA Voice Field Reference," on page 115. Call Progress Tone VoIP PIN Tone PSTN PIN Tone Outside Dial Tone Description This tone is
  • Cisco SPA2102-SF | Administration Guide - Page 101
    VoIP caller calls the FXO number, which auto-answers after VoIP Answer Delay. The ATA device then prompts the VoIP caller for a PIN. When a valid PIN is entered, the ATA plays the Outside Dial Tone and prompts the caller to dial the PSTN number. Cisco Small Business ATA Administration Guide 101
  • Cisco SPA2102-SF | Administration Guide - Page 102
    when the source IP address of the INVITE request is in the VoIP Access List . Using HTTP Digest Authentication (SPA3102) 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 ATA device PSTN
  • Cisco SPA2102-SF | Administration Guide - Page 103
    , 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. Call Forwarding to PSTN Gateway (SPA3102 and SPA8800
  • Cisco SPA2102-SF | Administration Guide - Page 104
    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
  • Cisco SPA2102-SF | Administration Guide - Page 105
    Network tabs in the administration web server pages. NOTE This information applies to the SPA2102, SPA3102, SPA8000, and SPA8800 models. To configure router settings for the PAP2T, and WRTP54G, see the user guide for the router. After you click the Router tab on the SPA2102, SPA3102, or the Network
  • Cisco SPA2102-SF | Administration Guide - Page 106
    . Current Gateway The default router assigned to the ATA device. Primary DNS The primary DNS server assigned to the ATA device. Secondary DNS The secondary DNS server assigned to the ATA device. LAN IP Address The address of the router. Cisco Small Business ATA Administration Guide 106
  • Cisco SPA2102-SF | Administration Guide - Page 107
    ," on page 111 Router tab > WAN Setup page > Internet Connection Settings section Connection Type The type of WAN connection. Options are: DHCP, Static IP, PPPoE, PPPoE / DHCP (tries PPPoE then DHCP), or DHCP/ PPPoE (tries DHCP then PPPoE). Cisco Small Business ATA Administration Guide 107
  • Cisco SPA2102-SF | Administration Guide - Page 108
    when DHCP is disabled. The default is 0.0.0.0. Router tab > WAN Setup page > PPPoE Settings section PPPoE Login Name The account name assigned by the ISP for connecting on a Point-to-Point Protocol over Ethernet (PPPoE) link. PPPoE Login Password The password assigned by the ISP for connecting
  • Cisco SPA2102-SF | Administration Guide - Page 109
    WAN Status page A Router tab > WAN Setup page > Optional Settings section HostName Domain Primary DNS The host name of the ATA device. The network domain of the ATA device. The DNS server that is used by the ATA device. Secondary DNS NOTE: When DHCP is enabled, you can enter the IP address
  • Cisco SPA2102-SF | Administration Guide - Page 110
    Yes. WAN Web Server Port The port that is used for WAN access to the ATA device. The default value is 80. Router tab > WAN Setup page > QOS Settings section Use Quality of Service (QoS) to assign different priority levels to different types of data transmissions. QOS Policy QOS QDisc Enable when
  • Cisco SPA2102-SF | Administration Guide - Page 111
    Lease Settings section," on page 112 Router tab > LAN Setup page > Networking Service section Networking Service NAT-the unit acts as a router and provides IP addresses to PCs attached to the LAN port. Bridge-The unit acts as a switch, a passthrough, and does not give IP addresses. Cisco Small
  • Cisco SPA2102-SF | Administration Guide - Page 112
    a static IP address. Enable Host Mac Address Host IP Address Options are Yes or No. Default is No. The device's MAC address The device's IP address Application page You can use the Application page to view the port forward settings and to update the DMZ settings, multicast passthru setting, and
  • Cisco SPA2102-SF | Administration Guide - Page 113
    section," on page 114 Router tab > Application page > Port Forwarding Settings section This feature allows you to set up specialized Internet applications that require port forwarding on a range of ports. Enable Service Name Starting Port Ending Port Protocol Server IP Address Options are Yes
  • Cisco SPA2102-SF | Administration Guide - Page 114
    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. That is, if there is a conflict - if port forwarding is set on the same port - then the port forwarding is cancelled. Default is 50000. Num
  • Cisco SPA2102-SF | Administration Guide - Page 115
    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. After you click the Voice tab, you can choose the following pages: • "Info page," on page 116 • "System page," on page
  • Cisco SPA2102-SF | Administration Guide - Page 116
    number. MAC address. Status of the client certificate, which can indicate if the ATA device has been authorized by your ITSP. For a Remote Configuration (RC) unit, this field indicates whether the unit has been customized or not. Pending indicates a new RC unit that is ready for provisioning. If the
  • Cisco SPA2102-SF | Administration Guide - Page 117
    Bytes Recv Total number of bytes of SIP messages received (including retransmissions). External IP External IP address used for NAT mapping. Voice tab > Info page > Line Status section (PSTN) Hook State Hook state of the FXO port. Options are either On or Off. Registration State Indicates if
  • Cisco SPA2102-SF | Administration Guide - Page 118
    the last caller. Mapped SIP Port Port number of the SIP port mapped by NAT. Call 1 and 2 State May take one of the following values: • Idle • Collecting PSTN Pin • Invalid PSTN PIN • PSTN Caller Accepted Call 1 and 2 Tone Call 1 and 2 Encoder Call 1 and 2 Decoder Call 1 and 2 FAX Call 1 and
  • Cisco SPA2102-SF | Administration Guide - Page 119
    . Error Call 1 and 2 The port mapped for Real Time Protocol traffic for Call 1/2. Mapped RTP Port Call 1 and 2 Media Media loopback is used to quantitatively and qualitatively Loopback measure the voice quality experienced by the end user. Cisco Small Business ATA Administration Guide 119
  • Cisco SPA2102-SF | Administration Guide - Page 120
    . Displays the default router assigned to the ATA device. Displays the primary DNS server assigned to the ATA device. Displays the secondary DNS server assigned to the ATA device. Voice tab > Info page > PSTN Line Status section (SPA3102) (PSTN) Hook State Hook state of the FXO port. Either On
  • Cisco SPA2102-SF | Administration Guide - Page 121
    some audio activity. Mapped SIP Port Port number of the SIP port mapped by NAT. Call Type May take one of the following values: • PSTN Gateway Call = VoIP-To-PSTN Call • VoIP Gateway Call = PSTN-To-VoIP Call • PSTN To Line 1 = PSTN call ring through and answered by Line 1 • Line 1 Forward to
  • Cisco SPA2102-SF | Administration Guide - Page 122
    VoIP Peer Number Phone number of the party at the VoIP call leg. PSTN Peer Number Phone number of the party at the PSTN call leg. VoIP Call Encoder Audio encoder being used for the VoIP call leg. VoIP Call Decoder Audio decoder being used for the VoIP call leg. VoIP Call FAX Status of the fax
  • Cisco SPA2102-SF | Administration Guide - Page 123
    You 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 survives after reboot or power cycle. Mapped SIP Port Port number of the SIP port mapped by NAT. Cisco Small Business ATA
  • Cisco SPA2102-SF | Administration Guide - Page 124
    page > System Configuration section Restricted Access This feature is used when implementing software Domains customization. Enable Web Server Enable/disable web server of the ATA device. This feature should only be used on firmware version 1.0.9 or later. The default is yes. Web Server Port
  • Cisco SPA2102-SF | Administration Guide - Page 125
    Sets the secondary DNS server to take over if problems are discovered with the Primary DNS server. This is in addition to DHCP-supplied DNS servers if DHCP is enabled; when DHCP is disabled, this is the secondary DNS server. The default is 0.0.0.0. Cisco Small Business ATA Administration Guide
  • Cisco SPA2102-SF | Administration Guide - Page 126
    is generated. To log SIP messages, Debug Level must be set to at least 2. Primary NTP Server Secondary NTP Server The default is 0. IP address or name of primary NTP server. IP address or name of secondary NTP server. Voice tab > System page > Miscellaneous Settings section (not used with
  • Cisco SPA2102-SF | Administration Guide - Page 127
    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. With some variations, depending on the model
  • Cisco SPA2102-SF | Administration Guide - Page 128
    ATA Voice Field Reference SIP page B 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
  • Cisco SPA2102-SF | Administration Guide - Page 129
    to yes, all AVT tone packets (encoded for redundancy) have the marker bit set. If set to no, only the first packet has the marker bit set for each DTMF event. SIP TCP Port Min SIP TCP Port Max The default is yes. Specifies the lowest TCP port number that can be used for SIP sessions. This field is
  • Cisco SPA2102-SF | Administration Guide - Page 130
    proxy in the Min-Expires header. If the value is larger than this setting, the maximum value is used. The default is 7200. Interval to wait before the ATA device retries registration after failing during the last registration. The default is 30. Cisco Small Business ATA Administration Guide 130
  • Cisco SPA2102-SF | Administration Guide - Page 131
    The default value is 0, which disables the exponential backoff feature. Voice tab > SIP page > Response Status Code Handling section SIT1 RSC SIT2 RSC SIT3 RSC SIP response status code for the appropriate Special Information Tone (SIT). For example, if you set the SIT1 RSC to 404, when the user
  • Cisco SPA2102-SF | Administration Guide - Page 132
    the SIT4 Tone. SIP response code that retries a backup server for the current request. 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
  • Cisco SPA2102-SF | Administration Guide - Page 133
    , not the time reported by an NTP server. If the ATA device receives a RR from the peer, it attempts to compute the round trip delay and show it as the value (ms) in the Info section of ATA device web page. The default is 0. No UDP Checksum Select yes if
  • Cisco SPA2102-SF | Administration Guide - Page 134
    page B Voice tab > SIP page > SDP Payload Types section NSE Dynamic Payload AVT Dynamic Payload INFOREQ Dynamic Payload G726r16 Dynamic Payload G726r24 Dynamic Payload G726r40 Dynamic Payload G729b Dynamic Payload NSE Codec Name NSE dynamic payload type. The valid range is 96-127. The default
  • Cisco SPA2102-SF | Administration Guide - Page 135
    . EncapRTP codec name used in SDP. The default is EncapRTP. Voice tab > SIP page > NAT Support Parameters section Handle VIA received If you select yes, the ATA device processes the received parameter in the VIA header (this value is inserted by the server in a response to anyone of its requests
  • Cisco SPA2102-SF | Administration Guide - Page 136
    this IP address when generating SIP messages and SDP (if NAT Mapping is enabled for that line). However, the results of STUN and VIA received parameter processing, if available, supersede this statically configured value. The default is 0.0.0.0. Cisco Small Business ATA Administration Guide 136
  • Cisco SPA2102-SF | Administration Guide - Page 137
    Voice Field Reference SIP page B EXT RTP Port Min NAT Keep Alive Intvl 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
  • Cisco SPA2102-SF | Administration Guide - Page 138
    ATA Voice Field Reference SIP page B 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
  • Cisco SPA2102-SF | Administration Guide - Page 139
    • "Ring and Call Waiting Tone Spec section" section on page 145 • "Control Timer Values (sec) section" section on page 146 • "Vertical Service Activation Codes section" section on page 148 • "Vertical Service Announcement Codes section (SPA2102, SPA8000)" section on page 154 • "Outbound Call Codec
  • Cisco SPA2102-SF | Administration Guide - Page 140
    the Dial Tone when the user dials a three-way call. The default is 420@-19,520@-19;10(*/0/1+2). Alternative to the Dial Tone. It prompts the user to enter an external phone number, as opposed to an internal extension. It is triggered by a, (comma) character encountered in the dial plan. The default
  • Cisco SPA2102-SF | Administration Guide - Page 141
    is configurable on the SIP screen. The default is 985@-16,1371@-16,1777@-16;20(.380/0/ 1,.274/0/2,.380/0/3,0/4/0). Played instead of the Dial Tone when there are unheard messages in the caller's mailbox. The default is 350@-19,440@-19;2(.1/.1/1+2);10(*/0/ 1+2). Played when all calls are forwarded
  • Cisco SPA2102-SF | Administration Guide - Page 142
    the VoIP gateway). This setting applies to the SPA3102 only. Feature Invocation Tone The default is 600@-10;*(0/.7/1,.2/.1/1,.2/.1/1,.2/.5/1). Played when a feature is implemented. The default is 350@-16;*(.1/.1/1). This field is not found in the PAP2T. Voice tab > Regional page > Distinctive
  • Cisco SPA2102-SF | Administration Guide - Page 143
    is 60(0.25/9.75). Cadence script for distinctive ring 9. This field is for the SPA2102 and SPA8000 only. The default is 60(.4/.2,.4/2). Voice tab > Regional page > Distinctive Call Waiting Tone Patterns section CWT1 Cadence CWT2 Cadence CWT3 Cadence CWT4 Cadence CWT5 Cadence Cadence script for
  • Cisco SPA2102-SF | Administration Guide - Page 144
    field is for the SPA2102 only. The default is 30(.3/9.7). Voice tab > Regional page > Distinctive Ring/CWT default is Bellcore-r5. Name in an INVITE's Alert-Info Header to pick distinctive ring/CWT 6 for the inbound call. The default is Bellcore-r6. Cisco Small Business ATA Administration Guide
  • Cisco SPA2102-SF | Administration Guide - Page 145
    call. This field is for the SPA2102 only. The default is Bellcore-r9. Voice tab > Regional page > Ring and Call Waiting Tone Spec section IMPORTANT: Ring and Call Waiting tones don't work the same way on all phones. When setting ring tones, consider the following recommendations: • Begin with
  • Cisco SPA2102-SF | Administration Guide - Page 146
    this tone. The default is 440@-10. Synchronized Ring If this is set to Yes, when a device calls the Linksys ATA, both lines ring at the same time (similar to a regular PSTN line). After one line answers, the other stops ringing. This field is only found in the PAP2T. No is the default. Voice tab
  • Cisco SPA2102-SF | Administration Guide - Page 147
    ) and the CPC feature should be used instead. Without CPC enabled, reorder tone will is played after a configurable delay. If CPC is enabled, dial tone will be played when tip-to-ring voltage is restored Resolution is 1 second. The default is 2. Cisco Small Business ATA Administration Guide 147
  • Cisco SPA2102-SF | Administration Guide - Page 148
    CPC is disabled if this value is set to 0. Range: 0 to 1.000 second. Resolution is 0.001 second. The default is 0 (CPC disabled). Voice tab > Regional page > Vertical Service Activation Codes section Vertical Service Activation Codes are automatically appended to the dial-plan. There is no need to
  • Cisco SPA2102-SF | Administration Guide - Page 149
    forwarding of all calls are enabled. Accept Last Deact Code CW Act Code The default is *64. Cancels the code to accept the last outbound call. The default is *84. Enables call waiting on all calls. CW Deact Code The default is *56. Disables call waiting on all calls. The default is *57. Cisco
  • Cisco SPA2102-SF | Administration Guide - Page 150
    Deact Code The default is *65. Disables caller ID generation. CWCID Act Code The default is *85. Enables call waiting, caller ID generation. CWCID Deact Code The default is *25. Disables call waiting, caller ID generation. The default is *45. Cisco Small Business ATA Administration Guide 150
  • Cisco SPA2102-SF | Administration Guide - Page 151
    ATA Voice Field Reference Regional page B Dist Ring Act Code Enables the distinctive ringing feature. Dist Ring Deact Code Speed Dial Act Code Secure All Call Act Code Secure No Call Act Code Secure One Call Act Code The default is *26 Disables the distinctive ringing feature. The default is *46
  • Cisco SPA2102-SF | Administration Guide - Page 152
    ATA Voice Field Reference Regional page B Referral Services Codes These codes tell the ATA device 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|*98|*123, etc. Max
  • Cisco SPA2102-SF | Administration Guide - Page 153
    ATA Voice Field Reference Regional page B 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
  • Cisco SPA2102-SF | Administration Guide - Page 154
    Codes section (SPA2102, SPA8000) Service Annc Base Number Service Annc Extension Codes Base number for service announcements. Extension codes for service announcements. Voice tab > Regional page > Outbound Call Codec Selection Codes section These codes automatically appended to the dial-plan
  • Cisco SPA2102-SF | Administration Guide - Page 155
    . The default is *0272640. Prefer G729a Code Makes this codec the preferred codec for the associated call. The default is *01729. Force G729a Code Makes this codec the only codec that can be used for the associated call. The default is *02729. Cisco Small Business ATA Administration Guide 155
  • Cisco SPA2102-SF | Administration Guide - Page 156
    . Sets the electrical impedance of the FXS port. Choices are 600, 900, 600+2.16uF, 900+2.16uF, 270+750||150nF, 220+850||120nF, 220+820||115nF, or 200+600||100nF. The default is 600. NOTE For New Zealand impedance (370+620||310nF), use 270+750||150nF. Cisco Small Business ATA Administration Guide
  • Cisco SPA2102-SF | Administration Guide - Page 157
    ATA Voice Field Reference Regional page B Daylight Saving Time Rule Enter the rule for calculating daylight saving time; it should include the start, end, and save (0-59). The default Daylight Saving Time Rule is start=4/1/ 7;end=10/-1/7;save=1. Cisco Small Business ATA Administration Guide 157
  • Cisco SPA2102-SF | Administration Guide - Page 158
    ATA Voice Field Reference Regional page B 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
  • Cisco SPA2102-SF | Administration Guide - Page 159
    ATA Voice Field Reference Regional page B Caller ID Method Caller ID FSK Standard FXS Port Power Limit Feature polarity reversal (and no DTAS) and before first ring. The default is Bellcore(N.Amer, China). The ATA device supports bell 202 and v.23 standards for caller ID generation. Select the
  • Cisco SPA2102-SF | Administration Guide - Page 160
    pages to configure the settings for the Line (FXO) ports 1 to 4. With some variations, depending on the model, this page includes the following sections: • "Line Enable section" section on page 161 • "Streaming Audio Server (SAS) section" section on page 162 • "NAT Settings section" section on page
  • Cisco SPA2102-SF | Administration Guide - Page 161
    • "Dial Plan section" section on page 180 • "FXS Port Polarity Configuration section" section on page 182 • "VoIP-to-PSTN Gateway Setup section (SPA8800)," on page 183 • "PSTN-To-VoIP Gateway Setup section (SPA8800)," on page 183 • "Call Forward, Speed Dial, Supplementary Services, and Ring Settings
  • Cisco SPA2102-SF | Administration Guide - Page 162
    Server (SAS) section On the SPA8800, these settings are configured on the Phone pages only. SAS Enable SAS DLG Refresh Intvl To enable the use of the line as a streaming audio source, select yes. Otherwise, select no. If enabled, the line cannot be used for outgoing calls. Instead, it auto-answers
  • Cisco SPA2102-SF | Administration Guide - Page 163
    value is empty. Voice tab > Line page > NAT Settings section NAT Mapping Enable NAT Keep Alive Enable NAT Keep Alive Msg To use externally mapped IP addresses and SIP/RTP ports in SIP messages, select yes. Otherwise, select no. The default is no. To send the configured NAT keep alive message
  • Cisco SPA2102-SF | Administration Guide - Page 164
    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. RTP ToS
  • Cisco SPA2102-SF | Administration Guide - Page 165
    are: UDP, TCP, TLS. The default is UDP. SIP Port Applies to SPA2102, SPA3102, and SPA8800. 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
  • Cisco SPA2102-SF | Administration Guide - Page 166
    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
  • Cisco SPA2102-SF | Administration Guide - Page 167
    ATA Voice Field Reference Line page B 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.
  • Cisco SPA2102-SF | Administration Guide - Page 168
    . 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. Voice tab > Line page > Call Feature Settings section On the SPA8800, these settings are configured on the Phone pages only. Blind Attn-Xfer Enable Enables the
  • Cisco SPA2102-SF | Administration Guide - Page 169
    Field Reference Line page B Conference Bridge Select the maximum number of conference call Ports participants. The range is 3 to 10. The default is 3. This field is found on the SPA2102 and PAP2T only. Voice tab > Line page or Phone page> Proxy and Registration section Proxy Outbound Proxy
  • Cisco SPA2102-SF | Administration Guide - Page 170
    1st proxy's port on the list. Voice Mail Server Mailbox Subscribe Expires The default is Normal. Enter the URL or IP address of the server. Expiry time to the voice mail server. The time to send another subscribe message to the voice mail server. Cisco Small Business ATA Administration Guide 170
  • Cisco SPA2102-SF | Administration Guide - Page 171
    Field Reference Line page B Voice tab > Line page > Subscriber Information section Display Name User ID Password Use Auth ID Display name for caller ID. Extension number to identify this line. Password for this line. To use the authentication ID and password for SIP authentication, select yes
  • Cisco SPA2102-SF | Administration Guide - Page 172
    the user has not subscribed for it, or b) the Service Provider intends to support similar service using other means than relying on the ATA device. On the SPA8800, these settings are configured on the Phone pages only. Call Waiting Serv Enable Call Waiting Service. Block CID Serv The default is
  • Cisco SPA2102-SF | Administration Guide - Page 173
    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. Cisco Small Business ATA Administration Guide 173
  • Cisco SPA2102-SF | Administration Guide - Page 174
    the Referral Services Codes parameter for more details. The default is yes. Enable Feature Dial Service. See the Feature Dial Services Codes parameter for more details. The default is yes. Enable Service Announcement Service. The default is yes. Voice tab > Line page > Audio Configuration section
  • Cisco SPA2102-SF | Administration Guide - Page 175
    fails if the far end does not support this codec.) Otherwise, select no. The default is no. Silence Supp Enable To enable silence suppression so that silent audio frames are not transmitted, select yes. Otherwise, select no. The default is no. Silence Threshold Select the appropriate setting
  • Cisco SPA2102-SF | Administration Guide - Page 176
    ATA Voice Field Reference Line page B 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 G.726 codec at 16 kbps, select yes. Otherwise, select no. The default is yes. Echo Supp Enable To
  • Cisco SPA2102-SF | Administration Guide - Page 177
    or SIP-INFO. The value can be set as low as 40 ms. There is no maximum limit. A larger value will reduce the chance of talk-off (beeping) during conversation, at the expense of reduced performance of DTMF detection, which is needed for interactive voice response systems (IVR). Default: 90 ms Cisco
  • Cisco SPA2102-SF | Administration Guide - Page 178
    Otherwise, select no. The default is yes. FAX Enable T38 To enable the use of the ITU-T T.38 standard for faxing, select yes. Otherwise, select no. T.38 is supported by the SPA2102 and the SPA8000. The SPA2102 supports a single T.38 connection. The SPA8000 supports one T.38 connection for each of
  • Cisco SPA2102-SF | Administration Guide - Page 179
    can be added to the dial plan such that when the user dials 1408+7digits, the call will be routed to Gateway 1. Without the syntax, all calls are routed to the given proxy by default (except IP dialing). The default is blank. Cisco Small Business ATA Administration Guide 179
  • Cisco SPA2102-SF | Administration Guide - Page 180
    value is the password to be used by the SPA to authenticate itself to Gateway 1. The default is blank. Voice tab > Line page > VoIP Fallback to PSTN section (SPA3102 and SPA8800) On the SPA8800, these settings are configured on the Phone pages only. Auto PSTN Fallback If enabled, the ATA device
  • Cisco SPA2102-SF | Administration Guide - Page 181
    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
  • Cisco SPA2102-SF | Administration Guide - Page 182
    ATA Voice Field Reference Line page B Enable IP Dialing Emergency Number Enable or disable IP dialing. If IP dialing is enabled, one can dial [user-id@]a.b.c.d[:port], where '@', '.', and ':' are dialed by entering *, user-id must be numeric (like a phone number) and a, b, c, d must be between 0
  • Cisco SPA2102-SF | Administration Guide - Page 183
    of the dial plan in the dial plan pool to be used when the VoIP Caller is calling from Line 1 of the same SPA8800 unit due to fallback to PSTN service when Line 1 VoIP service is down. Choose from {none, 1, 2, 3, 4, 5, 6, 7, 8}. The default is 1. Voice tab > Line page > PSTN-To-VoIP Gateway Setup
  • Cisco SPA2102-SF | Administration Guide - Page 184
    to add to the caller ID name. Voice tab > Line page > FXO Timer Values (sec) section (SPA8800) PSTN Answer Delay Delay in seconds before auto-answering inbound PSTN calls after the PSTN starts ringing. The range is 0-255. PSTN-To-VoIP Call Max Dur The default is 16. Limit on the duration of
  • Cisco SPA2102-SF | Administration Guide - Page 185
    SPA8800, these settings are configured on the Phone pages only. Detect CPC CPC is a brief removal of tip-and-ring voltage. If enabled, the SPA will disconnect both call legs when this signal is detected during a gateway call. The default is yes. Cisco Small Business ATA Administration Guide 185
  • Cisco SPA2102-SF | Administration Guide - Page 186
    The default is 0.2. If 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 Disconnect Tone parameter, which depends on the region of the PSTN service. The default is yes. Cisco Small Business
  • Cisco SPA2102-SF | Administration Guide - Page 187
    ATA Voice Field Reference Line page B Disconnect Tone Disconnect Tone continued 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
  • Cisco SPA2102-SF | Administration Guide - Page 188
    Field Reference Line page B Voice tab > Line page > International Control section (SPA8800) 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,
  • Cisco SPA2102-SF | Administration Guide - Page 189
    Services, and Ring Settings (SPA8000 and SPA8800) On the SPA8000 and the SPA8800, you can configure settings for call forward, speed dial, and supplementary services, as well as ring settings. For field descriptions, see "User page," on page 220. Cisco Small Business ATA Administration Guide
  • Cisco SPA2102-SF | Administration Guide - Page 190
    "SIP Settings section" section on page 191 • "Subscriber Information section" section on page 194 • "Dial Plan section" section on page 196 • "Proxy and Registration section" section on page 204 Voice tab > Trunk Group page (SPA8000)> Line Enable section Line Enable To enable this line for service
  • Cisco SPA2102-SF | Administration Guide - Page 191
    Internet browsing or e-commerce. Options are: UDP, TCP, TLS. The default is UDP. Port number of the SIP message listening and transmission port. SIP 100REL Enable The default is 5060. To enable the support of 100REL SIP extension for reliable transmission of provisional responses (18x) and use of
  • Cisco SPA2102-SF | Administration Guide - Page 192
    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
  • Cisco SPA2102-SF | Administration Guide - Page 193
    ATA Voice Field Reference Trunk Group page (SPA8000) B 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
  • Cisco SPA2102-SF | Administration Guide - Page 194
    page (SPA8000) B Voice tab > Trunk Group page > (SPA8000) > Subscriber Information section Display Name User ID Password Use Auth ID Auth ID Call Capacity Display name for caller ID. Extension number for this line. Password for this line. To use the authentication ID and password for SIP tone to
  • Cisco SPA2102-SF | Administration Guide - Page 195
    assigned to this particular trunk. • You can instruct the SPA8000 to hunt only the phones that are on-hook, through the Voice tab > SIP page, Trunking Parameters section, Hunt Policy field. See "Trunking Parameters section (SPA8000)," on page 138. Cisco Small Business ATA Administration Guide 195
  • Cisco SPA2102-SF | Administration Guide - Page 196
    By default the trunk dial plan allows any non-empty number: ([*#0-9AD][*#0-9A-D].) Voice tab > Trunk Group page (SPA8000) > NAT Settings section NAT Mapping Enable NAT Keep Alive Enable NAT Keep Alive Msg NAT Keep Alive Dest To use externally mapped IP addresses and SIP/RTP ports in SIP messages
  • Cisco SPA2102-SF | Administration Guide - Page 197
    B Voice tab > Trunk Group page (SPA8000) > Proxy and Registration section Proxy Use Outbound Proxy Outbound Proxy Use OB Proxy In Dialog Register Make Call Without Reg Register Expires Ans Call Without Reg SIP proxy server for all outbound requests. Enablse the use of an Outbound Proxy. If set to
  • Cisco SPA2102-SF | Administration Guide - Page 198
    server. PSTN Line page (SPA3102) On the SPA3102, you can use the Voice tab > PSTN Line page to configure your PSTN line. This page includes the following sections: • "Line Enable section" section on page 161 • "NAT Settings section" section on page 199 Cisco Small Business ATA Administration Guide
  • Cisco SPA2102-SF | Administration Guide - Page 199
    Phone 1, or Phone 2. The default is Phone1+2. Voice tab > PSTN Line page (SPA3102) > NAT Settings section NAT Mapping Enable To use externally mapped IP addresses and SIP/RTP ports in SIP messages, select yes. Otherwise, select no. The default is no. Cisco Small Business ATA Administration Guide
  • Cisco SPA2102-SF | Administration Guide - Page 200
    NAT keep alive messages. If the value is $PROXY, the messages are sent to the current or outbound proxy. The default is $PROXY. Voice tab > PSTN Line page (SPA3102) > Network Settings section SIP ToS/DiffServ Value TOS/DiffServ field value in UDP IP packets carrying a SIP message. The default
  • Cisco SPA2102-SF | Administration Guide - Page 201
    only, or disable. The default is up and down. Voice tab > PSTN Line page (SPA3102) > SIP Settings section 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
  • Cisco SPA2102-SF | Administration Guide - Page 202
    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
  • Cisco SPA2102-SF | Administration Guide - Page 203
    ATA Voice Field Reference PSTN Line page (SPA3102) B 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
  • Cisco SPA2102-SF | Administration Guide - Page 204
    ) B Voice tab > PSTN Line page (SPA3102) > Proxy and Registration section Proxy Use Outbound Proxy Outbound Proxy Use OB Proxy In Dialog Register Make Call Without Reg Register Expires Ans Call Without Reg SIP proxy server for all outbound requests. Enable the use of Outbound Proxy. If set to no
  • Cisco SPA2102-SF | Administration Guide - Page 205
    on the 1st proxy's port on the list. The default is Normal. Voice tab > PSTN Line page (SPA3102) > Subscriber Information section Display Name User ID Password Display name for caller ID. Extension number for this line. Password for this line. Cisco Small Business ATA Administration Guide 205
  • Cisco SPA2102-SF | Administration Guide - Page 206
    on this line interface. Choices: {unlimited,1,2,3,...25 }. Default is 16. Note that the ATA device does not distinguish between incoming and outgoing calls when talking about call capacity. NOTE: unlimited = 16 Voice tab > PSTN Line page (SPA3102) > Audio Configuration section A codec resource
  • Cisco SPA2102-SF | Administration Guide - Page 207
    is yes. To enable detection of the fax Caller-Entered Digits (CED) tone, select yes. Otherwise, select no. G726-32 Enable The default is yes. To enable the use of the G726 codec at 32 kbps, select yes. Otherwise, select no. The default is yes. Cisco Small Business ATA Administration Guide 207
  • Cisco SPA2102-SF | Administration Guide - Page 208
    ATA Voice Field Reference PSTN Line page (SPA3102) B FAX CNG Detect Enable To enable detection of the fax Calling Tone (CNG), select yes. Otherwise, select no. G726-40 Enable The default is yes. To enable the use of the G726 codec at 40 kbps, select yes. Otherwise, select no. FAX Passthru
  • Cisco SPA2102-SF | Administration Guide - Page 209
    = 16). INFO uses SIP INFO with the single line signal=hf in the message body. The MIME type for this message body is taken from the Hook Flash MIME Type setting. FAX Disable ECAN The default is None. If enabled, this feature automatically disables the echo canceller when a fax tone is detected. To
  • Cisco SPA2102-SF | Administration Guide - Page 210
    Dial Plan 1 to 8. The dial plan syntax is the same as that used for Line 1. See "Dial Plan section," on page 180. Voice tab > PSTN Line page (SPA3102) > VoIP-To-PSTN Gateway Setup section VoIP-To-PSTN Gateway Enable Enable or disable VoIP-To-PSTN Gateway functionality. The default is yes. VoIP
  • Cisco SPA2102-SF | Administration Guide - Page 211
    The default is blank. VoIP Caller 1/2/3/4/ One of 8 PIN numbers that can be specified to control 5/6/7/8 PIN access to the PSTN gateway by a VoIP Caller, when the VoIP Caller Authentication Method parameter is set to PIN. The default is blank. Cisco Small Business ATA Administration Guide 211
  • Cisco SPA2102-SF | Administration Guide - Page 212
    the dial plan in the dial plan pool to be associated 5/6/7/8 DP with the VoIP caller who enters the PIN that matches VoIP Caller 1/2/3/4/5/6/7/8 PIN. The default is 1. Voice tab > PSTN Line page (SPA3102) > VoIP Users and Passwords (HTTP Authentication) section VoIP User 1/2/3/4/ 5/6/7/8 Auth
  • Cisco SPA2102-SF | Administration Guide - Page 213
    Reference PSTN Line page (SPA3102) B Voice tab > PSTN Line page (SPA3102) > Ring Settings section Default Ring 1-8, Follow Line Cfg Voice tab > PSTN Line page (SPA3102) > FXO (PSTN) Timer Values (sec) section VoIP Answer Delay Delay in seconds before auto-answering inbound VoIP calls for the
  • Cisco SPA2102-SF | Administration Guide - Page 214
    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. The permitted range is limited to 0.02 to 1.6 seconds. Cisco Small Business ATA Administration Guide
  • Cisco SPA2102-SF | Administration Guide - Page 215
    PSTN gateway call, the 1st polarity reversal is ignored and the 2nd one triggers the disconnection. For VoIP gateway call, the 1st polarity reversal triggers the disconnection. The default is yes. Detect (PSTN) Long Silence If enabled, SPA will disconnect both call legs when the PSTN side has no
  • Cisco SPA2102-SF | Administration Guide - Page 216
    Disconnect Tone If 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 Disconnect Tone parameter, which depends on the region of the PSTN service. (PSTN) Long Silence Duration The default is
  • Cisco SPA2102-SF | Administration Guide - Page 217
    ATA Voice Field Reference PSTN Line page (SPA3102) B 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
  • Cisco SPA2102-SF | Administration Guide - Page 218
    Field Reference PSTN Line page (SPA3102) B Voice tab > PSTN Line page (SPA3102) > 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
  • Cisco SPA2102-SF | Administration Guide - Page 219
    -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. Cisco Small Business ATA Administration Guide 219
  • Cisco SPA2102-SF | Administration Guide - Page 220
    device shall use the user and line settings for that line; there is no user login support. Per user parameter tags must be appended with [1] or [2] (corresponding to line 1 or 2) in the configuration profile. It is omitted below for readability. Cisco Small Business ATA Administration Guide 220
  • Cisco SPA2102-SF | Administration Guide - Page 221
    Reference User page B 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
  • Cisco SPA2102-SF | Administration Guide - Page 222
    to Cfwd Last Dest by using the Call Forward Last activation code Cfwd Last Dest The default is blank. Forward number for the Cfwd Last Caller parameter. Same as Cfwd All Dest. The default is blank. Voice tab > User page > Speed Dial Settings section This section does not apply to the WIP310
  • Cisco SPA2102-SF | Administration Guide - Page 223
    calls are secure calls by default. Message Waiting The default is no. This value is updated when there is voice mail notification received by the ATA device. The 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
  • Cisco SPA2102-SF | Administration Guide - Page 224
    and Packet. Default is Media. Note that if the ATA device answers the call, then the loopback type is determined by the caller (the ATA device always picks the first loopback type in the offer if it contains multiple types.) Voice tab > User page > Distinctive Ring Settings section Caller number
  • Cisco SPA2102-SF | Administration Guide - Page 225
    User page B Voice tab > User page > Ring Settings section Default Ring Default ringing pattern, 1 - 8, for all callers. Default CWT The default is 1. Default CWT pattern, 1 - 8, for all callers. Hold Reminder Ring The default is 2. Ring pattern for reminder of a holding call when the phone
  • Cisco SPA2102-SF | Administration Guide - Page 226
    Voice tab > PSTN User page (SPA3102) > PSTN-To-VoIP Selective Call Forward Settings section Cfwd Sel1-8 Caller Cfwd Sel1-8 Dest Eight PSTN Caller Number Patterns to be blocked for VoIP gateway services or forwarded to a certain VoIP number. If the caller is blocked, the SPA will not auto-answers
  • Cisco SPA2102-SF | Administration Guide - Page 227
    ATA Voice Field Reference PSTN User page (SPA3102) B Voice tab > PSTN User page (SPA3102) > PSTN-To-VoIP Speed Dial Settings section Speed Dial 1-9 The VoIP number to call when the PSTN caller dials a single digit '2' Voice tab > PSTN User page (SPA3102) > PSTN Ring Thru Line 1 Distinctive Ring
  • Cisco SPA2102-SF | Administration Guide - Page 228
    ATA Voice Field Reference PSTN User page (SPA3102) B Cisco Small Business ATA Administration Guide 228
  • Cisco SPA2102-SF | Administration Guide - Page 229
    Write down the IP address as it is announced. 5. Press 7932#. 6. Press 1 to enable WAN access to the administration web server. 7. Open the web browser on a networked computer. 8. Start Internet Explorer and enter the IP address of the ATA device. Cisco Small Business ATA Administration Guide 229
  • Cisco SPA2102-SF | Administration Guide - Page 230
    3. You do not need to change the value of the syslog server parameter. 4. To capture SIP signaling messages, under the Line tab, set SIP Debug Option to Full. The file output is syslog..log (for the default port setting, syslog.514.log). Cisco Small Business ATA Administration Guide 230
  • Cisco SPA2102-SF | Administration Guide - Page 231
    DHCP. (For help with port forwarding, consult your router documentation) 2. On the Line tab of the administration web server, change the value of Nat Mapping Enable to yes. On the SIP tab; change Substitute VIA Addr to yes, and the EXT IP parameter to the IP address of your router. 3. Make sure you
  • Cisco SPA2102-SF | Administration Guide - Page 232
    web server, set STUN Enable to yes, and enter the IP address of the STUN server in STUN Server. STUN (Simple Traversal of UDP through NATs) is a protocol defined by RFC 3489, that allows a client behind a NAT device to find out its public address, the type of NAT it is behind, and the port
  • Cisco SPA2102-SF | Administration Guide - Page 233
    page 233 • "SPA2102," on page 234 • "SPA3102," on page 234 • "SPA8000," on page 235 • "SPA8800," on page 235 • "WRTP54G," on page 236 Device 3.98" x 3.98" x 1.10" (101 x 101 x 28 mm) W x H x D Dimensions Unit Weight 5.40 oz (153g) Power Cisco Small Business ATA Administration Guide 233
  • Cisco SPA2102-SF | Administration Guide - Page 234
    Environmental Specifications SPA2102 D SPA2102 Device Dimensions Unit Weight Power Certification Operating Temp Storage Temp Operating Humidity Storage Humidity 3.98" x 3.98" x 1.10" ( Non-Condensing 10% to 90% relative humidity, Non-Condensing Cisco Small Business ATA Administration Guide 234
  • Cisco SPA2102-SF | Administration Guide - Page 235
    Non-Condensing 10% to 90% relative humidity, Non-Condensing SPA8800 Device Dimensions Unit Weight Power Certification Operating Temp Storage Temp Operating Humidity Storage Humidity 6.69" x 1.54" x 8. 10% to 90% relative humidity, Non-Condensing Cisco Small Business ATA Administration Guide 235
  • Cisco SPA2102-SF | Administration Guide - Page 236
    Environmental Specifications WRTP54G WRTP54G Device Dimensions Unit Weight Power Certification Operating Temp Storage Temp Operating Humidity Storage Humidity 6.69 " x humidity, Non-Condensing 5% to 90% relative humidity, Non-Condensing D Cisco Small Business ATA Administration Guide 236
  • Cisco SPA2102-SF | Administration Guide - Page 237
    , you may have to add http:// in front of the web address. Resource Location Technical Documentation http://www.cisco.com/en/US/products/ps10024/ prod_maintenance_guides_list.html Firmware Downloads Go to tools.cisco.com/support/downloads, and enter the model number in the Software Search box
  • Cisco SPA2102-SF | Administration Guide - Page 238
    . End Users, VARs, and Service Providers SPA9000 Voice System Installation and Configuration Guide Web-UI (Legacy) Based Product Configuration Manual installation of the SPA9000 Voice System, by using the Web User Interface, instead of the Cisco SPA900 Voice System Setup Wizard. End Users, VARs
  • Cisco SPA2102-SF | Administration Guide - Page 239
    • SPA9x2 series IP phones Analog Telephone Adapter Administration Guide • Administration and use of Cisco Small Business ATAs • PAP2T, SPA2102, SPA3102, SPA8000, and WRTP54G VARs, system administrators, and Service Providers User Guide for switch User Guide for router Cisco Small Business ATA
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Cisco Small Business
SPA2102, SPA3102, SPA8000, SPA8800, PAP2T
Analog Telephone Adapters
ADMINISTRATION
GUIDE