Table of Contents

Installation

Mounting Instructions and GUI Overview

Quick Setup for Voice over IP Service

Configuring the Network

Configuring the Voice Settings

Administration Settings

Viewing the Status and Statistics

Frequently Asked Questions

Using the IVR for Administration

Advanced Options for Voice Services

Where to Go From Here

Getting Started

Product Features

Top Panel

Feature

Description

Feature

Description

Reset

Using a paperclip or similar object, press this button briefly to restart the unit. Press and hold for 10 seconds to restore the factory default settings.

Phone 1, Phone 2 (Gray)

Connect to an analog phone, using an RJ-11 phone cable.

Ethernet (Yellow)

SPA122 Only

Can be used to connect to a device on your network, such as a computer, using an Ethernet cable.

Internet (Blue)

Connect to a broadband network device (DSL or cable modem) or a network router, using an Ethernet cable.

Power

Connect to a power source, using the provided power adapter.

Connecting the Equipment

NOTE For wall-mounting instructions, see Mounting the ATA.

STEP 1 Connect one end of the provided Ethernet cable to the Internet (Blue) port. Connect the other end directly to your broadband network device.

STEP 2 Connect one end of a phone cable to the Phone 1 (Gray) port. Connect the other end to your analog phone or fax machine.

STEP 3 Connect one end of another phone cable to another analog phone or fax machine. Connect the other end to the Phone 2 (Gray) port.

STEP 4 SPA122 Only: Optionally, connect one end of an Ethernet network cable to the ETHERNET (Yellow) port of the ATA. Connect the other end to a device on your network, such as a computer.

STEP 5 Connect the provided power adapter to the Power port.

Configuration and Management of the ATA

You can use the web-based configuration utility to set up your ATA. You also can use the built-in Interactive Voice Response (IVR) system.

Using the Web-Based Configuration Utility

STEP 1 Connect your computer to the same subnet as the ATA. For example, if the ATA is connected to a LAN port on your router, also connect your computer to a LAN port on your router.

Note: On SPA122, you can connect your computer to the ETHERNET (Yellow) port of the ATA.

STEP 2 Power on your computer.

NOTE: Make sure your computer’s Ethernet adapter is set to obtain an IP address automatically. For more information, refer to the Help for your operating system.

STEP 3 Start a web browser on your computer.

STEP 4 In the Address bar, enter the IP address of the ATA.

- SPA112: Use the ATA’s IVR or your router’s configuration utility to find the dynamically assigned IP address of the ATA. For information about the IVR, see Using the IVR for Administration.

- SPA122: In the Address bar, enter: 192.168.15.1

Note: 192.168.15.1 is the default local IP address of the ATA.

STEP 5 To log in for the first time, enter the default username, admin, and the default password, admin. The password is case sensitive.

STEP 6 Enter the Connection Type and settings required by your Internet Service Provider. Types include DHCP (the default option), Static IP, and PPPoE (required for most DSL service). After entering these settings, click Submit to establish your Internet connection.

STEP 7 Use the menus to configure your settings, as needed.

Using the IVR for Administration

An IVR system is available to help you to configure and manage your ATA. You can use the telephone keypad to select options and to make your entries.

To access the IVR menu:

STEP 1 Connect an analog phone to the Phone port of the ATA.

STEP 2 Press the star key four times: ****

STEP 3 After the greeting plays, press the keys on the phone keypad to select your options.

STEP 4 Enter the code for the desired action. See the IVR Actions table for details.

TIPS:

- Enter the numbers slowly, listening for the audio confirmation before entering the next number.

- After you select an option, press the # (pound) key.

- To exit the menu, hang up the telephone or enter 3948# to exit.

- After entering a value, such as an IP address, press the # (pound) key to indicate that you have finished your selection. To save the new setting, press 1 . To review the new setting, press 2 . To re-enter the new setting, press 3 . To cancel your entry and return to the main menu, press * (star).

- While entering a value, you can cancel the changes by pressing the * (star) key twice within half a second. Be sure to press the key quickly, or the * will be treated as a decimal point entry.

- If the menu is inactive for more than one minute, the ATA times out. You will need to re-enter the menu by pressing the star key four times: **** . Your settings take effect after you hang up the telephone or exit the IVR. The ATA may reboot at this time.

- To enter the decimal points in an IP address, press the * (star) key. For example, to enter the IP address 191.168.1.105, perform the following tasks:

– Press these keys: 191*168*1*105.

– Press the # (pound) key to indicate that you have finished entering the IP address.

– Press 1 to save the IP address or press the * (star) key to cancel your entry and return to the main menu.

IVR Actions

IVR Action

Menu Option

Choices and Instructions

Enter IVR Menu

****

Check Internet Connection Type

100

101

DHCP: 0

Static IP: 1

PPPoE: Press 2

PPPoE, DHCP: Press 3

DHCP, PPPoE: Press 4

Check Internet IP Address (WAN port)

110

111

Enter the IP address by using numbers on the telephone key pad. Use the * (star) key when entering a decimal point.

Note: This option is available only after you choose Static IP as the Internet Connection Type, through option 101.

Check Network Mask

Set Network Mask

121

To enter the value, press numbers on the telephone key pad. Press the * (star) key to enter a decimal point.

Note: This option is available only after you choose Static IP as the Internet Connection Type, through option 101.

Check Gateway IP Address

130

Set Gateway IP Address

131

To enter the value, press numbers on the telephone key pad. Press the * (star) key to enter a decimal point.

Note: This option is available only after you choose Static IP as the Internet Connection Type, through option 101.

Check MAC Address

140

150

Set Primary DNS Server

161

To enter the value, press numbers on the telephone key pad. Press the * (star) key to enter a decimal point.

Note: This option is available only after you choose Static IP as the Internet Connection Type, through option 101.

Check Internet web server port

170

Announce Line 1 SIP Transport

1910

1911

0: UDP

1: TCP

2: TLS

Check Line 2 SIP Transport

1920

1921

0: UDP

1: TCP

2: TLS

Exit IVR

3948

7932

1: Enable

0: Disable

Factory Reset of Unit

WARNING: All non-default settings will be lost. This includes network and service provider data.

73738

RESET

When prompted, press 1 to confirm, or press * (star) to cancel. After you hear “Option successful,” hang up the phone. The ATA reboots.

Reboot of Voice System

732668

REBOOT

After you hear “Option successful,” hang up the phone. The ATA reboots.

Mounting the ATA

You can place the ATA on a desktop or mount it on a wall.

CAUTION To prevent the ATA from overheating, do not operate it in an area that exceeds an ambient temperature of 104°F (40°C).

Desktop Placement

Place the ATA on a flat surface near an electrical outlet.

WARNING Do not place anything on top of the ATA; excessive weight could damage it.

Wall Mounting

The ATA has two wall-mount slots on the bottom panel. To mount the ATA on a wall, you need mounting hardware (not included). Suggested hardware is illustrated (not true to scale).

Recommended hardware (not included): Two number-six pan-head tapping screws, 5/8-in. length, with anchors for sheet rock installation.

5/8 in. (15.8 mm)

WARNING Insecure mounting might damage the ATA or cause injury. Cisco is not responsible for damages incurred by insecure wall-mounting.

To mount the unit to the wall:

STEP 1 Determine where you want to mount the unit. Verify that the surface is smooth, flat, dry, and sturdy.

STEP 2 Drill two pilot holes into the surface 58 mm apart (about 2.28 in.).

STEP 3 Insert a screw into each hole, leaving a gap of 5 mm (0.1968 in.) between the underside of each screw head and the surface of the wall.

STEP 4 Place the unit wall-mount slots over the screws and slide the unit down until the screws fit snugly into the wall-mount slots.

Elements of the User Interface

Component

Description

1. Menu Bar

(top)

Provides access to the modules of the configuration utility. Click a menu to view the options in the navigation tree.

2. Navigation Tree

(left panel)

Provides access to the configuration pages within the selected module. Click a category heading to view the list of features. Click a link to open the configuration page.

3. Configuration Page

(main area)

Settings for the selected feature.

Configuration Utility Icons

Icon

Description

Edit Icon

The Edit icon lets you edit an existing item from a list. After making your changes, click the Submit button to save your changes.

Add Item Icon

The Add Item icon lets you add an item to a list. After you have created a new item, click the Submit button to save the new item.

Delete Item Icon

The Delete Item icon lets you delete an item from a list. After you have deleted an item, click the Submit button to save your changes.

Saving the Settings

Your settings on a configuration page are not saved until you click the Submit button. When you navigate to another page, any unsaved settings are abandoned. Changes cannot be saved while calls are in progress. Try again when the phones are idle.

To clear the settings without saving them, you can click the Cancel button.

Help

To view information about the configuration pages, click the Help link near the top right corner of the configuration utility. You can then use the table of contents to find topics of interest.

Logout

Quick Setup for Voice over IP Service

The Quick Setup page is displayed automatically when you first log on ATA. You can use this page to quickly configure connectivity to your provider’s Voice over IP network for your analog phones.

NOTE Connecting to your service provider’s network requires Internet connectivity. With the default network settings, your ATA should have Internet connectivity when you connect a cable from the WAN port of the ATA to a port on your router or broadband network device. For more information, see Internet Settings.

STEP 1 For Line 1 and Line 2, specify the settings for the phone services to be used by a analog (FXS) phones or fax machines that are connected to the PHONE1 and PHONE2 ports.

- Proxy: Enter the domain name or URL of the service provider’s proxy server.

- Display Name: Enter the name that you want to use to identify your account. This name typically is used as your Caller ID name.

- User ID: Enter the user ID that is required to log in to your Internet account.

- Password: Enter the password that is required to log in to your Internet account.

- Dial Plan in ( Line section only) : Keep the default settings (recommended) or edit the dial plan to suit your site. For more information, see Configuring Dial Plans.

STEP 2 Click Submit to save your settings. The voice service will restart.

STEP 3 To verify your progress, perform the following tasks:

a. Click Voice in the menu bar, and then click Info in the navigation tree. Verify that the Registration State is Registered for Line 1 and Line 2 Status.

If the line is not registered, you may need to refresh the browser several times because it can take a few seconds for the registration to complete. Also verify that your Internet Settings, including DNS server settings, are configured according to the information from your ISP. For more information, see Internet Settings.

b. Use an external phone to place an inbound call to the telephone number that was assigned by your ITSP. Verify that the phone rings and you have two-way audio on the call.

Configuring the Network

This chapter describes how to configure the network settings for your ATA. It includes the following sections:

- Basic Setup

- Advanced Settings

- Application (SPA122 Only)

Basic Setup

Use the Network Setup > Basic Setup pages to configure your Internet connection, local network settings (SPA122 only), and your time settings.

- Network Service (SPA122 Only)

- Internet Settings

- Network Settings for the LAN and DHCP Server (SPA122 Only)

- Time Settings

Network Service (SPA122 Only)

Use the Network Setup > Basic Setup > Network Service page to configure the operating mode of the SPA122.

After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

You can configure the ATA to operate in one of the following modes:

- NAT: Network Address Translation (NAT) is a function that allows multiple devices on a private network to share a public, routable IP address to establish connections over the Internet. To enable Voice over IP service to co-exist with NAT, some form of NAT traversal is required, either on the ATA or another network device. Use this option if your ATA connects to one network on the WAN port (10.0.0.0 for example) and to another network on the LAN port (192.168.0.0 for example). This option is selected by default and is suitable for most deployments.

- Bridge: Bridged mode is used if the ATA is acting as a bridge device to another router. Choose this option if your ATA bridges a network (10.0.0.0 for example) to its LAN port (with connected devices also in the 10.0.0x range).

Internet Settings

Use the Network Setup > Basic Setup > Internet Settings page to set up your Internet connection.

Enter the settings as described in the table. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

Internet Connection Type

Field

Description

Connection Type

Specify the Internet addressing method that your ISP requires. Default setting: Automatic Configuration - DHCP

- Automatic Configuration - DHCP: Use this setting if your ISP dynamically provides an IP address. No additional settings are required on this page.

- Static IP: Use this setting if your ISP assigned a static/permanent IP address. Complete the fields that appear. See more information below.

- PPPoE (DSL service): Some DSL-based ISPs use PPPoE (Point-to-Point Protocol over Ethernet) to establish Internet connections. If you are connected to the Internet through a DSL line, check with your ISP to see if they use PPPoE. Complete the fields that appear. See more information below.

Static IP Settings

- Internet IP Address and Subnet Mask: Enter the IP address and subnet mask that was assigned to your account by your service provider. This address is seen by external users on the Internet.

- Default Gateway: Enter the Gateway IP Address that was provided by your ISP.

If needed, you can adjust the MTU and Optional Settings, as described below.

PPPoE Settings

- User Name and Password: Enter the user name and password that you use to log into your ISP network through a PPPoE connection.

- Service Name: If provided by your ISP, enter the Service Name.

- Connect on Demand: You can configure the ATA to disconnect your Internet connection after a specified period of inactivity (Max Idle Time). If your Internet connection has been terminated due to inactivity, this feature enables the ATA to automatically re-establish your connection as soon as you attempt to access the Internet again. If you choose this option, also set the Max Idle Time.

- Keep Alive: This option keeps you connected to the Internet indefinitely, even when your connection sits idle. If you choose this option, also set the Redial Period, which is the interval at which the ATA verified Internet connectivity. The default period is 30 seconds.

If needed, you can adjust the MTU and Optional Settings, as described below.

MTU

The Maximum Transmission Unit (MTU) setting specifies the largest protocol data unit (in bytes) permitted for network transmission. Generally, a larger MTU means greater efficiency. However, a larger packet may cause delays for other traffic and is more likely to become corrupted. In most cases, you should keep the default setting, Auto, to allow the ATA to choose the appropriate MTU. To specify the MTU, select Manual, and then enter the number of bytes.

Optional Settings

Feature

Description

Host Name

The name of the ATA. The default value is the model number. Your ISP may specify a host name to use.

Domain Name

The domain name, if specified by your ISP. Otherwise, leave the field blank.

DNS Server Order

Choose the preferred method for choosing a DNS server.

- DHCP-Manual— The DNS server settings from the network server will take precedence, and your entries in the DNS fields below will be used only as a backup.

- Manual-DHCP— Your entries in the DNS fields below will take precedence, and the DNS server settings from the network server will be used as a backup.

- Manual— Your entries in the DNS fields below will be used to choose a DNS server.

Primary DNS

Enter the IP address of the primary Domain Name Service (DNS) server to use for domain name resolution. Keep the default entry, 0.0.0.0, to use the primary DNS server that is specified for the WAN connection.

Secondary DNS

Enter the IP address of the secondary Domain Name Service (DNS) server to use for domain name resolution. Keep the default entry, 0.0.0.0, to use the secondary DNS server that is specified for the WAN connection.

Network Settings for the LAN and DHCP Server (SPA122 Only)

Use the Network Setup > Basic Setup > Network Settings page to set the IP address and subnet mask for your local network. Also configure the settings for the built-in DHCP server (SPA122 only).

Enter the settings as described below. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

Router IP

Enter the Local IP Address and Subnet Mask for your local network. The default setting is 192.168.15.1 with a subnet mask of 255.255.255.0.

DHCP Server Setting

Field

Description

DHCP Server

The ATA can use the built-in DHCP server to dynamically assign IP addresses to connected devices. Click Enabled to enable the DHCP server, or click Disabled to disable this feature.

Default setting: Enabled

IP Reservation:

Click the Show DHCP Reservation button to view and manage the DHCP client list. Click the Hide DHCP Reservation button to hide the list. When the list is displayed, you can perform the following tasks:

- To reserve a static IP address for a current DHCP client: Check the box for the client in the Select Clients from DHCP Tables list. Click Add Clients . The selected clients are added to the Clients Already Reserved list. These clients have static IP addresses that do not change.

- To add a client that is not in the Select Clients from DHCP Tables list: Type a name for the client in the Enter Client Name box. Enter an IP address for this client in the Assign IP Address box. Enter the MAC address in the following format: 00:00:00:00:00:00. Click Add .

- To remove a client from the Clients Already Reserved list: Check the box for the client. Click Remove .

Default Gateway

Enter the IP address of the default gateway to be used by the DHCP clients.

Default setting: 192.168.15.1 (the IP address of the ETHERNET (LAN) interface)

Starting IP Address

Enter the first address in the range of addresses to be assigned dynamically by the DHCP server.

Default setting: 192.168.15.100

Maximum DHCP Users

Enter the maximum number of devices that can dynamically receive, or “lease,” DHCP addresses from the DHCP server.

Default setting: 50

IMPORTANT: Typically, the ATA can support up to five connected computers for business-related tasks such as web browsing and viewing email. The ATA is not designed to support streaming music, video, games, or other network traffic-intensive tasks.

Client Lease Time

Enter the number of minutes that a dynamically assigned IP address can be in use, or “leased.” After this time elapses, a client device has to request a DHCP lease renewal. Use 0 to represent 1 day, 9999 never expire.

Default setting: 0

Option 66

Provides provisioning server address information to hosts that request this option. Server information can be defined in one of three ways:

- None: The ATA uses its own TFTP server to source provisioning files, so it returns its own local IP address to the client.

- Remote TFTP Server: The ATA was configured by using this method, and received server information through Option 66 on its WAN interface. In response to client requests, it provides the remote TFTP server information.

- Manual TFTP Server: Allows the manual configuration of a configuration server address. While this option is typically used to provide either an IP address or a fully qualified hostname, the ATA will also accept and offer a full URL including protocol, path and filename to meet to requirements of specific clients.

Default setting: None

TFTP Server

If you chose Manual TFTP Server for Option 66, enter the IP address, hostname, or URL of the TFTP server that is used to configure the ATA.

Default setting: blank

Option 67

Provides a configuration/bootstrap filename to hosts that request this option. This option is used in conjunction with option 66 to allow a client to form an appropriate TFTP request for the file.

Default setting: blank

Option 159

Provides a configuration URL to clients that request this option. An option 159 URL defines the protocol and path information by using an IP address for clients that cannot use DNS. For example:

https://10.1.1.1:888/configs/bootstrap.cfg

Default setting: blank

Option 160

Provides a configuration URL to clients that request this option. An option 160 URL defines the protocol and path information by using a fully qualified domain name for clients that can use DNS. For example:

https://myconfigs.cisco.com:888/configs/bootstrap.cfg

Default setting: blank

DNS Proxy

When enabled, the DNS proxy relays DNS requests to the current public network DNS server for the proxy, and replies as a DNS resolver to the client device on the network. Click Enabled to enable this feature, or click Disabled to disable it. If DNS proxy is disabled, then DHCP clients will be offered DNS server information by using the Static DNS servers, if defined, or by using the using the servers specified for the INTERNET (WAN) interface.

Default setting: Enabled

Time Settings

Use the Network Setup > Basic Setup > Time Settings page to set the system time for the ATA. By default, the system time is set automatically by using a Network Time Protocol (NTP) server. You can configure the system time manually. In addition, you can use this page to specify your time zone, enable Daylight Saving adjustments, and modify related settings.

After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

User Manual

If you prefer to set the system manually rather than automatically obtaining the settings from an NTP server, click User Manual and then enter the date and time.

Field

Description

Date

Enter the date in the following order: four-digit year, month, day.

Time

Enter the time in the following order: hour (from 1 to 24), minutes, and seconds.

Time Zone

To use a time server to establish the time settings, select Time Zone. Then complete the fields in this section.

Field

Description

Time Zone

Choose the time zone for the site where the ATA is in operation.

Default setting: (GMT-08:00) Pacific Time (USA & Canada)

Adjust Clock for Daylight Saving Changes

Check the box if you want to automatically adjust the time when Daylight Savings Time is in effect. Otherwise, uncheck the box.

Default setting: Enabled

Time Server Address

To use the ATA’s default Network Time Protocol (NTP) server, select Auto from the drop-down list. This is the default setting. If you want to specify the NTP server, select Manual, and then enter the NTP server address.

Default setting: Auto

Resync Timer

Enter the Resync timer interval value (in seconds). This timer controls how often the ATA resynchronizes with the NTP server.

Default setting: 3600 seconds

Auto Recovery After Reboot

Choose this option to allow the ATA to automatically reconnect to the time server after a system reboot.

Default setting: Disabled

Advanced Settings

Use the Network Setup > Advanced Settings pages to configure features including port flow control, MAC address cloning, VPN passthrough, and VLAN.

- Port Setting (SPA122 Only)

- MAC Address Clone (SPA122 Only)

- VPN Passthrough (SPA122 Only)

- VLAN

- CDP & LLDP

Port Setting (SPA122 Only)

Use the Network Setup > Advanced Settings > Port Setting page to set the ETHERNET (LAN) port attributes.

After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

Field

Description

Flow Control

Flow control is a mechanism that temporarily stops the transmission of data on a port. For example, a situation might arise where a sending station (computer) is transmitting data faster than some other part of the network (including the receiving station) can accept. The overwhelmed network element will halt the transmission of the sender for a specified period of time.

Choose Enabled to enable this feature, or choose Disabled to disable this feature.

Default setting: Enabled

Speed Duplex

Choose the duplex mode. You can select from Auto-negotiate, 10 Half, 10 Full, 100 Half and 100 Full. Cisco recommends choosing Auto-negotiate to automatically select the appropriate mode for the traffic. Use caution with other settings. Problems can result if you choose a setting that is not appropriate for the network devices.

Default setting: Auto-negotiate

MAC Address Clone (SPA122 Only)

A MAC address is a 12-digit code assigned to a unique piece of hardware for identification purposes. Some ISPs require that you register a MAC address in order to access the Internet. If you previously registered your account with another MAC address, it may be convenient to assign that MAC address to your ATA. You can use the Network Setup > Advanced Settings > MAC Address Clone page to assign a MAC address that you previously registered with your Service Provider.

After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

Field

Description

MAC Clone

Click Enabled to enable MAC address cloning, or click Disabled to disable this feature.

Default setting: Disabled.

MAC Address

Enter the MAC address that you want to assign to your ATA. If your computer’s MAC address is the address that you previously registered for your ISP account, click Clone Your PC’s MAC. Your computer’s MAC address appears in the MAC Address field.

Default setting: the current MAC address of your ATA

VPN Passthrough (SPA122 Only)

Use the Network Setup > Advanced Settings > VPN Passthrough page to configure VPN passthrough for IPsec, PPTP, and L2TP protocols. Use this feature if there are devices behind the ATA that need to set up IPsec tunnels independently. For example, a device may need to use a VPN tunnel to connect to another router on the WAN.

By default, VPN Passthrough is enabled for IPsec, PPTP, and L2TP.

After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

Field

Description

IPsec Passthrough

Internet Protocol Security (IPsec) is a suite of protocols used to implement secure exchange of packets at the IP layer. Click Enabled to enable this feature, or click Disabled to disable it. Default setting: Enabled

PPTP Passthrough

Point-to-Point Tunneling Protocol (PPTP) allows the Point-to-Point Protocol (PPP) to be tunneled through an IP network. To disable PPTP Passthrough, select Disabled. Default setting: Enabled

L2TP Passthrough

Layer 2 Tunneling Protocol is the method used to enable Point-to-Point sessions via the Internet on the Layer 2 level. Click Enabled to enable this feature, or click Disabled to disable it. Default setting: Enabled

VLAN

Use the Network Setup > Advanced Settings > VLAN page to assign a VLAN ID to your network. For example, your call control system may require a particular voice VLAN ID.

After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

Field

Description

Enable VLAN

Click Enabled to enable a VLAN, or click Disabled to disable this feature. Default setting: Disabled

VLAN ID

The VLAN ID can be any numeral from 1 through 4094. When VLAN is enabled, the default setting is 1.

CDP & LLDP

Device discovery protocols enable directly connected devices to discover information about each other. You may wish to enable these protocols to allow your network management system to learn about your ATA and endpoints. Use the Network Setup > Advanced Settings > CDP & LLDP page to specify the settings for Cisco Discovery Protocol (CDP) and the Link Layer Discovery Protocol (LLDP). When a discovery protocol is enabled, the ATA sends periodic messages to a multicast address and also listens to the periodic messages sent by other devices that use that protocol.

After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

Application (SPA122 Only)

Use the Network Setup > Application pages to support voice service and any servers that you host for public access.

- Quality of Service (QoS) (SPA122 Only)

- Port Forwarding (SPA122 Only)

- DMZ (SPA122 Only)

Quality of Service (QoS) (SPA122 Only)

Use the Network Setup > Application > QoS page to set the upstream bandwidth to suit your broadband service. This feature is enabled by default and helps to ensure that voice is prioritized during periods of heavy network traffic.

Enter the settings as described below. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

Field

Description

QoS Policy

Click Always On to enable QoS settings at all times, or click On When Phone In Use to enable it only when there is voice traffic.

Default setting: On When Phone In Use

Upstream Bandwidth

Enter the maximum available upstream bandwidth value specified by your Internet Service Provider.

Default setting: 10000 kbps

IMPORTANT: Do not overstate the upstream bandwidth that you receive from your service provider. Setting this value higher than the available service bandwidth can result in traffic being dropped arbitrarily in the service provider's network.

Port Forwarding (SPA122 Only)

Use the Network Setup > Application > Port Forwarding page if you need to explicitly allow access to specific ports from external devices.

List of Port Forwarding

To add a port forwarding rule, click Add Entry. To edit a port forwarding rule, select it in the list and then click the pencil icon. To remove a port forwarding rule, click the delete icon. For more information, see Manually Adding Port Forwarding (SPA122 Only).

Field

Description

Number

An identification number for the port forwarding rule.

Type

The type of rule: Single Port Forwarding or Port Range Forwarding.

Status

The status of the rule: Enabled or Disabled.

Application

The application that uses this rule to access a network resource.

Port Forwarding Details

To display the details, click an entry in the List of Port Forwarding.

Field

Description

External Port

The port that external clients will use to set up this connection.

Internal Port

The port that the ATA uses when forwarding traffic to the internal server.

Protocol

The protocol that is used: TCP or UDP.

IP Address

The IP address of the internal server that is accessed by this rule.

Manually Adding Port Forwarding (SPA122 Only)

Use this page to enter the port forwarding settings for an application.

Enter the settings as described below. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

Field

Description

Port Forwarding Type

Choose the type of port forwarding:

- Single Port Forwarding : Forwards traffic for a specified port to the same or an alternative port on the target server in the LAN.

- Port Range Forwarding : Forwards traffic to a range of ports to the same ports on the target server in the LAN. see the Internet application’s documentation for the required ports or ranges.

Application Name

For single port forwarding, choose a common application from the drop-down list (such as Telnet, or DNS).

To add an application that is not on the list, choose Add a new name, and then enter the name in the Enter a Name field.

Enter a Name

If you chose Port Range Forwarding, or if you chose Add a new name in the Application Name list for Single Port Forwarding, enter a name to identify the application.

External Port, Internal Port

For Single Port Forwarding, specify the ports to use. For simplicity, the internal and external port numbers will often be the same. However, different external port numbers could be used to differentiate traffic of the same application type intended for different internal servers, or to promote privacy through the use of non-standard ports.

- External port: For single port forwarding, enter the port number that external clients will use to set up a connection with the internal server.

- Internal port: For single port forwarding, enter the port number that the ATA uses when forwarding traffic to the internal server.

The correct entries appear automatically if you choose a standard application from the Application Name list for Single Port Forwarding.

Start - End Port

For Port Range Forwarding, specify the range of ports to use. Valid values are from 1 to 65535.

Protocol

Select the protocol(s) that can be forwarded: TCP, UDP, or TCP and UDP.

IP Address

Enter the IP address of the local server that will receive forwarded traffic.

To ensure correct forwarding of traffic, local servers must either be configured with a static IP address, or be assigned a reserved IP address through DHCP. Use the Interface Setup > LAN > DHCP Server page to reserve IP addresses. See Network Settings for the LAN and DHCP Server (SPA122 Only).

Enabled

Check the box to enable this port forwarding rule, or uncheck the box to disable it. Default setting: Disabled

DMZ (SPA122 Only)

Use the Network Setup > Application > DMZ page if you need to allow a local device to be exposed to the Internet for a special-purpose service.

The specified network device must have its DHCP client function disabled and must have a reserved IP address (also known as a static IP address) to ensure that it is reachable at the specified IP address. To reserve an IP address for a device, see Network Settings for the LAN and DHCP Server (SPA122 Only).

NOTE A Demilitarized Zone (DMZ) is similar to Port Range Forwarding. Both features allow Internet traffic to access a resource on your private network. However, Port Range Forwarding is more secure because it only opens the ports that you specify for an application. DMZ hosting opens all the ports of one device, exposing it to the Internet.

Enter the settings as described below. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

Field

Description

Status

Click Enabled to enable this feature, or click Disabled to disable it. Default setting: Disabled

Private IP

Specify the local IP address of the device that can be accessed through the DMZ.

Configuring the Voice Settings

This chapter describes how to configure the voice settings and voice services for the ATA. It includes the following sections:

- Information

- System

- SIP

- Provisioning

- Regional

- Line 1 and Line 2 Settings (PHONE 1 and PHONE 2)

- User 1 and User 2

NOTE For additional information, see Appendix 1, “Advanced Options for Voice Services.”

Information

Use the Voice > Information page to view information about the ATA voice application.

Product Information

Field

Description

Product Name

Model number/name.

Serial Number

Product serial number.

Software Version

Software version number.

Hardware Version

Hardware version number.

MAC Address

MAC Address. For example: 8843E1657936.

Client Certificate

Status of the client certificate, which can indicate if the ATA was authorized by your ITSP.

Customization

Used for Remote Configuration by service providers who deploy the ATA to their customers.

- Open: Not a Remote Configuration unit. This ATA can be configured by using the configuration utility.

- Pending: This Remote Configuration unit has not yet connected to the server to get its profile.

- Customized Value: This Remote Configuration unit has received its profile from the server. For Example: ISP_Name

System Status

Field

Description

Current Time

Current date and time of the system; for example, 10/3/2003 16:43:00. Set the system time by using the Network Setup > Time Settings page.

Elapsed Time

Total time elapsed since the last reboot of the system; for example, 25 days and 18:12:36.

RTP Packets Sent

Total number of RTP packets sent (including redundant packets)

RTP Bytes Sent

Total number of RTP bytes sent.

RTP Packets Recv

Total number of RTP packets received (including redundant packets)

RTP Bytes Recv

Total number of RTP bytes received.

SIP Messages Sent

Total number of SIP messages sent (including retransmissions)

SIP Bytes Sent

Total number of bytes of SIP messages sent (including retransmissions)

SIP Messages Recv

Total number of SIP messages received (including retransmissions)

SIP Bytes Recv

Total number of bytes of SIP messages received (including retransmissions)

External IP

The External IP address used for NAT mapping.

Line 1/Line 2 Status

Field

Description

Hook State

The hook state of the port: On or Off.

Registration State

Indicates if the line has registered with the SIP proxy.

Last Registration At

Last date and time the line was registered.

Next Registration In

The number of seconds before the next registration renewal.

Message Waiting

Indicates whether you have new voice mail waiting.

Indicates Yes when a message is received.

Mapped SIP Port

Port number of the SIP port mapped by NAT.

Call Back Active

Indicates whether or not a call back request is in progress. Options are either yes or no.

Last Called Number

The phone number that was most recently called through this port.

Last Caller Number

The originating phone number of the call that was most recently received through this port.

Call 1 and 2 State

Indicates the state of calls.

Call 1 and 2 Tone

The type of tone used by the call.

Call 1 and 2 Encoder

The codec used for encoding.

Call 1 and 2 Decoder

The codec used for decoding.

Call 1 and 2 FAX

The status of the fax mode.

Call 1 and 2 Type

The direction of the call. May take one of the following values:

- Inbound

- Outbound

Call 1 and 2 Remote Hold

Indicates whether the far end has placed the call on hold.

Call 1 and 2 Callback

Indicates whether the call was triggered by a call back request.

Call 1 and 2 Peer Name

The name of the peer phone.

Call 1 and 2 Peer Phone

The phone number of the peer phone.

Call 1 and 2 Duration

The duration of the call.

Call 1 and 2 Packets Sent

The number of packets sent

Call 1 and 2 Packets Recv

The number of packets received.

Call 1 and 2 Bytes Sent

The number of bytes sent.

Call 1 and 2 Bytes Recv

The number of bytes received.

Call 1 and 2 Decode Latency

The number of milliseconds for decoder latency.

Call 1 and 2 Jitter

The number of milliseconds for receiver jitter

Call 1 and 2 Round Trip Delay

The number of milliseconds for delay.

Call 1 and 2 Packets Lost

The number of packets lost.

Call 1 and 2 Packet Error

The number of invalid packets received.

Custom CA Status

Field

Description

Custom CA Provisioning Status

The status of the latest custom CA (Certificate Authority) certificate download.

Custom CA Info

The successfully downloaded CA information, or “Not Installed” if no custom CA certificate was installed.

Default setting: Not Installed

System

Use the Voice > System page to configure general voice system settings and to enable logging by using a syslog server. (Logging also can be configured in the Administration > Logging pages. For more information, see Logging.)

Requirements for Logging

- You need a computer that is on the same subnetwork as the ATA, to capture the log files. This computer needs to be running a syslog daemon. Enter the IP address and the port of this computer in the Syslog Server and Debug Server fields. Default port is 514.

- You can deploy a syslog server to receive syslog messages from the ATA, which acts as a syslog client. The syslog client device uses the syslog protocol to send messages, based on its configuration, to a syslog server. The syslog messages can be accessed by reviewing the "syslog.514.log" file which is the default name resides in the same directory as the slogsrv.exe syslog server application.

- Partners can download the Syslog Server for SPA Devices by using the link below (login required):

www.cisco.com/en/US/partner/prod/collateral/voicesw/ps6788/phones/ps10499/syslog_server_for_spa_devices.zip

Enter the settings as described below. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

System Configuration

Field

Description

Restricted Access Domains

Domain that Cisco IP phones responds to SIP messages only from the identified servers. Applicable to Line 1.

IVR Admin Passwd

Password for the administrator to manage the ATA by using the built-in IVR through a connected phone.

Network Startup Delay

The number of seconds of delay between restarting the voice module and initializing network interface.

Default setting: 3

Miscellaneous Settings

Field

Description

DNS Query TTL Ignore

In DNS packages, the server will suggest a TTL value to the client; if this parameter is set to yes, the value from the server will be ignored.

Default setting: no

Syslog Server

Specify the syslog server domain name/IP and port. This feature specifies the server for logging ATA system information and critical events. If both Debug Server and Syslog Server are specified, Syslog messages are also logged to the Debug Server.

Default setting: blank

Debug Server

The debug server domain name/IP and port. This feature specifies the server for logging debug information. The level of detailed output depends on the debug level parameter setting.

Default setting: blank

Debug Level

Determines the level of debug information that will be generated. Select 0, 1, 2, 3, 3+Router, 3+Coma or 1+Coma from the drop-down list. The higher the debug level, the more debug information will be generated. Level 0 means that no information will be collected. Levels 1, 2 & 3 generate messages related to the voice ports only. Level 3+Router generates debug content for both voice and router components.

Levels 1/3+ Coma generates vendor’s cordless debug content and voice component.

Default setting: 3

Syslog Prefix

Prefixes additional information to syslog with configuration set.

Default setting: blank

SIP

Use the Voice > SIP page to configure SIP parameters and values.

Enter the settings as described below. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

NOTE For a deeper understanding of these fields, refer to Request for Comments (RFC) 3261.

SIP Parameters

Field

Description

Max Forward

The maximum times a call can be forwarded. The valid range is from 1 to 255.

Default setting: 70

Max Redirection

Number of times an invite can be redirected to avoid an infinite loop.

Default setting: 5.

Max Auth

The maximum number of times (from 0 to 255) a request may be challenged.

Default setting: 2

SIP User Agent Name

The User-Agent header used in outbound requests. If empty, the header is not included. Macro expansion of $A to $D corresponding to GPP_A to GPP_D allowed.

Default setting: $VERSION

SIP Server Name

The server header used in responses to inbound responses.

Default setting: $VERSION

SIP Reg User Agent Name

The User-Agent name to be used in a REGISTER request. If this value is not specified, the SIP User Agent Name parameter is also used for the REGISTER request.

Default setting: blank

SIP Reg Starting Sequence Number

Defines the SIP Reg message Sequence Number.

Default setting: blank

SIP Accept Language

Accept-Language header used. There is no default (this indicates that the ATA does not include this header) If empty, the header is not included.

Default setting: blank

DTMF Relay MIME Type

The MIME Type used in a SIP INFO message to signal a DTMF event.

Default setting: application/dtmf-relay.

Hook Flash MIME Type

The MIME Type used in a SIP INFO message to signal a hook flash event.

Default setting: application/hook-flash

Remove Last Reg

Determines whether or not the ATA removes the last registration before submitting a new one, if the value is different. Select yes to remove the last registration, or select no to omit this step.

Default setting: no

Use Compact Header

Determines whether or not the ATA uses compact SIP headers in outbound SIP messages. Select yes or no from the drop-down list. Select yes to use compact SIP headers in outbound SIP messages. Select no to use normal SIP headers. If inbound SIP requests contain compact headers, the ATA reuses the same compact headers when generating the response regardless the settings of the Use Compact Header parameter. If inbound SIP requests contain normal headers, the ATA substitutes those headers with compact headers (if defined by RFC 261) if Use Compact Header parameter is set to yes.

Default setting: no

Escape Display Name

Determines whether or not the Display Name is private. Select yes if you want the ATA to enclose the string (configured in the Display Name) in a pair of double quotes for outbound SIP messages. If the display name includes " or \, these will be escaped to \" and \\ within the double quotes. Otherwise, select no.

Default setting: no

RFC 2543 Call Hold

Configures the type of call hold: a:sendonly or 0.0.0.0. Do not use the 0.0.0.0 syntax in a HOLD SDP; use the a:sendonly syntax.

Default setting: yes

Mark all AVT Packets

Select yes if you want all AVT tone packets (encoded for redundancy) to have the marker bit set for each DTMF event. Select no to have the marker bit set only for the first packet.

Default setting: yes

AVT Packet Size

Indicates the AVT Packet size according to value set in ptime or fixed 10ms.

Default setting: ptime

SIP TCP Port Min

The lowest TCP port number that can be used for SIP sessions.

Default setting: 5060

SIP TCP Port Max

The highest TCP port number that can be used for SIP sessions.

Default setting: 5080

CTI Enable

Enables or disables the Computer Telephone Interface feature provided by some servers.

Default setting: no

Keep Referee When REFER Failed

Set this parameter to yes to configure the phone to immediately handle NOTIFY sipfrag messages.

You can also configure this parameter in the configuration file:

<Keep_Referee_When_REFER_Failed ua="na">Yes

</Keep_Referee_When_REFER_Failed>

Caller ID Header

Provides the option to take the caller ID from PAID-RPID-FROM, P-ASSERTEDIDENTITY, REMOTE-PARTY-ID, or FROM header.

Default setting: PAID-RPID-FROM

SIP Timer Values

Field

Description

SIP T1

RFC 3261 T1 value (round-trip time estimate), which can range from 0 to 64 seconds.

Default setting: 0.5

SIP T2

RFC 3261 T2 value (maximum retransmit interval for non-INVITE requests and INVITE responses), which can range from 0 to 64 seconds.

Default setting: 4

SIP T4

RFC 3261 T4 value (maximum duration a message remains in the network), which can range from 0 to 64 seconds.

Default setting: 5

SIP Timer B

INVITE time-out value, which can range from 0 to 64 seconds.

Default setting: 32

SIP Timer F

Non-INVITE time-out value, which can range from 0 to 64 seconds.

Default setting: 16

SIP Timer H

H INVITE final response, time-out value, which can range from 0 to 64 seconds.

Default setting: 32

SIP Timer D

ACK hang-around time, which can range from 0 to 64 seconds.

Default setting: 32

SIP Timer J

Non-INVITE response hang-around time, which can range from 0 to 64 seconds.

Default setting: 32

INVITE Expires

INVITE request Expires header value. If you enter 0, the Expires header is not included in the request. Range: 0–(231–1)

Default setting: 240

ReINVITE Expires

ReINVITE request Expires header value. If you enter 0, the Expires header is not included in the request. Range: 0–(231–1)

Default setting: 30

Reg Min Expires

Minimum registration expiration time allowed from the proxy in the Expires header or as a Contact header parameter. If the proxy returns a value less than this setting, the minimum value is used.

Default setting: 1

Reg Max Expires

Maximum registration expiration time allowed from the proxy in the Min-Expires header. If the value is larger than this setting, the maximum value is used.

Default setting: 7200

Reg Retry Intvl

Interval to wait before the ATA retries registration after failing during the last registration.

Default setting: 30

Reg Retry Long Intvl

When registration fails with a SIP response code that does not match Retry Reg RSC, the ATA waits for the specified length of time before retrying. If this interval is 0, the ATA stops trying. This value should be much larger than the Reg Retry Intvl value, which should not be 0.

Default setting: 1200

Reg Retry Random Delay

Random delay range (in seconds) to add to Register Retry Intvl when retrying REGISTER after a failure.

Default setting: 0 (disabled)

Reg Retry Long Random Delay

Random delay range (in seconds) to add to Register Retry Long Intvl when retrying REGISTER after a failure.

Default setting: 0 (disabled)

Reg Retry Intvl Cap

The maximum value to cap the exponential back-off retry delay (which starts at Register Retry Intvl and doubles on every REGISTER retry after a failure) In other words, the retry interval is always at Register Retry Intvl seconds after a failure. If this feature is enabled, Reg Retry Random Delay is added on top of the exponential back-off adjusted delay value.

Default setting: 0, which disables the exponential backoff feature.

Response Status Code Handling

Field

Description

SIT1 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 makes a call and a failure code of 404 is returned, the SIT1 tone is played. Reorder or Busy tone is played by default for all unsuccessful response status code for SIT 1 RSC through SIT 4 RSC.

Default setting: blank

SIT2 RSC

SIP response status code to INVITE on which to play the SIT2 Tone.

Default setting: blank

SIT3 RSC

SIP response status code to INVITE on which to play the SIT3 Tone.

Default setting: blank

SIT4 RSC

SIP response status code to INVITE on which to play the SIT4 Tone.

Default setting: blank

Try Backup RSC

SIP response code that retries a backup server for the current request.

Default setting: blank

Retry Reg RSC

Interval to wait before the ATA retries registration after failing during the last registration.

Default setting: blank

RTP Parameters

Field

Description

RTP Port Min

Minimum port number for RTP transmission and reception.

The RTP Port Min and RTP Port Max parameters should define a range that contains at least 4 even number ports, such as 100 –106.

Default setting: 16384.

RTP Port Max

Maximum port number for RTP transmission and reception.

Default setting: 16482.

RTP Packet Size

Packet size in seconds, which can range from 0.01 to 0.16. Valid values must be a multiple of 0.01 seconds.

Default setting: 0.030

RTP Tx Packet Size Follows Remote SDP

Enable the Remote pair RTP Packet Size.

Default setting: Yes

Max RTP ICMP Err

Number of successive ICMP errors allowed when transmitting RTP packets to the peer before the ATA terminates the call. If value is set to 0, the ATA ignores the limit on ICMP errors.

Default setting: 0

RTCP Tx Interval

Interval for sending out RTCP sender reports on an active connection. It can range from 0 to 255 seconds. During an active connection, the ATA can be programmed to send out compound RTCP packet on the connection. Each compound RTP packet except the last one contains a SR (Sender Report) and a SDES (Source Description) The last RTCP packet contains an additional BYE packet. Each SR except the last one contains exactly 1 RR (Receiver Report); the last SR carries no RR. The SDES contains CNAME, NAME, and TOOL identifiers. The CNAME is set to <User ID>@<Proxy>, NAME is set to <Display Name> (or Anonymous if user blocks caller ID), and TOOL is set to the Vendor/Hardware-platform-software-version. The NTP timestamp used in the SR is a snapshot of the local time for the ATA, not the time reported by an NTP server. If the ATA receives a RR from the peer, it attempts to compute the round trip delay and show it as the Call Round Trip Delay value (ms) on the Information page.

Default setting: 0

No UDP Checksum

Select yes if you want the ATA to calculate the UDP header checksum for SIP messages. Otherwise, select no.

Default setting: no

Stats In BYE

Determines whether the ATA includes the P-RTP-Stat header or response in a BYE message. The header contains the RTP statistics of the current call. Select yes or no from the drop-down list.

Default setting: yes

The format of the P-RTP-Stat header is:

P-RTP-State: PS=<packets sent>,OS=<octets sent>,PR=<packets received>,OR=<octets received>,PL=<packets lost>,JI=<jitter in ms>,LA=<delay in ms>,DU=<call duration ins>,EN=<encoder>,DE=<decoder>.

SDP Payload Types

Field

Description

NSE Dynamic Payload

NSE dynamic payload type. The valid range is 96-127.

Default setting: 100

AVT Dynamic Payload

AVT dynamic payload type. The valid range is 96-127.

Default setting: 101

INFOREQ Dynamic Payload

INFOREQ dynamic payload type.

Default setting: blank

G726r32 Dynamic Payload

G726r32 dynamic payload type.

Default setting: 2

G729b Dynamic Payload

G.729b dynamic payload type. The valid range is 96-127.

Default setting: 99

EncapRTP Dynamic Payload

EncapRTP Dynamic Payload type.

Default setting: 112

RTP-Start-Loopback Dynamic Payload

RTP-Start-Loopback Dynamic Payload type.

Default setting: 113

RTP-Start-Loopback Codec

RTP-Start-Loopback Codec. Select one of the following: G711u, G711a, G726-32, G729a.

Default setting: G711u

NSE Codec Name

NSE codec name used in SDP.

Default setting: NSE

AVT Codec Name

AVT codec name used in SDP.

Default setting: telephone-event

G711u Codec Name

G.711u codec name used in SDP.

Default setting: PCMU

G711a Codec Name

G.711a codec name used in SDP.

Default setting: PCMA

G726r32 Codec Name

G.726-32 codec name used in SDP.

Default setting: G726-32

G729a Codec Name

G.729a codec name used in SDP.

Default setting: G729a

G729b Codec Name

G.729b codec name used in SDP.

Default setting: G729ab

EncapRTP Codec Name

EncapRTP codec name used in SDP.

Default setting: encaprtp

NAT Support Parameters

Field

Description

Handle VIA received

If you select yes, the ATA processes the received parameter in the VIA header (this value is inserted by the server in a response to any one of its requests) If you select no, the parameter is ignored. Select yes or no from the drop-down menu.

Default setting: no

Handle VIA rport

If you select yes, the ATA processes the rport parameter in the VIA header (this value is inserted by the server in a response to any one of its requests) If you select no, the parameter is ignored. Select yes or no from the drop-down menu.

Default setting: no

Insert VIA received

Inserts the received parameter into the VIA header of SIP responses if the received-from IP and VIA sent-by IP values differ. Select yes or no from the drop-down menu.

Default setting: no

Insert VIA rport

Inserts the parameter into the VIA header of SIP responses if the received-from IP and VIA sent-by IP values differ. Select yes or no from the drop-down menu.

Default setting: no

Substitute VIA Addr

Lets you use NAT-mapped IP:port values in the VIA header. Select yes or no from the drop-down menu.

Default setting: no

Send Resp To Src Port

Sends responses to the request source port instead of the VIA sent-by port. Select yes or no from the drop-down menu.

Default setting: no

STUN Enable

Enables the use of STUN to discover NAT mapping. Select yes or no from the drop-down menu.

Default setting: no

STUN Test Enable

If the STUN Enable feature is enabled and a valid STUN server is available, the ATA can perform a NAT-type discovery operation 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 ATA detects symmetric NAT or a symmetric firewall, NAT mapping is disabled.

Default setting: no

STUN Server

IP address or fully-qualified domain name of the STUN server to contact for NAT mapping discovery.

Default setting: blank

EXT IP

External IP address to substitute for the actual IP address of the ATA in all outgoing SIP messages. If 0.0.0.0 is specified, no IP address substitution is performed.

If this parameter is specified, the ATA assumes 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.

This option requires that you have (1) a static IP address from your Internet Service Provider and (2) an edge device with a symmetric NAT mechanism. If the ATA is the edge device, the second requirement is met.

Default setting: blank

EXT RTP Port Min

External port mapping number of the RTP Port Min. number. If this value is not zero, the RTP port number in all outgoing SIP messages is substituted for the corresponding port value in the external RTP port range.

Default setting: blank

NAT Keep Alive Intvl

Interval between NAT-mapping keep alive messages.

Default setting: 15

Redirect Keep Alive

Enables or disables NAT Redirect keep alive messages.

Default setting: no

Linksys Key System Parameters

Field

Description

Linksys Key System

To enable operation with the Cisco SPA9000, choose yes. Otherwise, choose no.

Default setting: no

Multicast Address

The multicast address for devices in the Cisco SPA9000 voice network.

Default setting: 224.168.168.168:6061

Key System Auto Discovery

To enable auto-discovery of the Cisco SPA9000 voice system, choose yes. Otherwise, choose no.

Default setting: yes

Key System IP Address

The IP address of the Cisco SPA9000.

Default setting: blank

Force LAN Codec

If needed, specify a voice codec.

Default setting: none

Station Name

Name to identify this IP phone; appears on the IP phone screen on models that have a display. No spaces are allowed and the name must be unique.

Default setting: blank

Provisioning

Use the Voice > Provisioning page to configure profiles and parameters to provision the ATA from a remote server.

Enter the settings as described below. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

Configuration Profile

Field

Description

Provision Enable

Controls all resync actions independently of firmware upgrade actions. Set to yes to enable remote provisioning.

Default setting: yes

Resync On Reset

Triggers a resync after every reboot except for reboots caused by parameter updates and firmware upgrades.

Default setting: yes

Resync Random Delay

The maximum value for a random time interval that the ATA waits before making its initial contact with the provisioning server. This delay is effective only on the initial configuration attempt following power-on or reset. The delay is a pseudo-random number between zero and this value.

This parameter is in units of 20 seconds; the default value of 2 represents 40 seconds. This feature is disabled when this parameter is set to zero.

This feature can be used to prevent an overload of the provisioning server when a large number of devices power-on simultaneously.

Default setting: 2 (40 seconds)

Resync At (HHmm)

The time of day when the device tries to resync. The resync is performed each day. Used in conjunction with the Resync At Random Delay.

Default setting: blank

Resync At Random Delay

Used in conjunction with the Resync At (HHmm) setting, this parameter sets a range of possible values for the resync delay. The system randomly chooses a value from this range and waits the specified number of seconds before attempting to resync. This feature is intended to prevent the network jam that would occur if all resynchronizing devices began the resync at the exact same time of day.

Default setting: 600

Resync Periodic

The time interval between periodic resyncs with the provisioning server. The associated resync timer is active only after the first successful synchronization with the server. Setting this parameter to zero disables periodic resynchronization.

Default setting: 3600

Resync Error Retry Delay

Resync retry interval (in seconds) applied in case of resync failure.

The ATA has an error retry timer that activates if the previous attempt to sync with the provisioning server fails. The ATA waits to contact the server again until the timer counts down to zero.

This parameter is the value that is initially loaded into the error retry timer. If this parameter is set to zero, the ATA immediately retries to sync with the provisioning server following a failed attempt.

Default setting: 3600

Forced Resync Delay

Maximum delay (in seconds) that the ATA waits before performing a resync.

The ATA does not resync while one of its lines is active. Because a resync can take several seconds, it is desirable to wait until the ATA has been idle for an extended period before resynchronizing. This allows a user to make calls in succession without interruption.

The ATA has a timer that begins counting down when all of its lines become idle. This parameter is the initial value of the counter. Resync events are delayed until this counter decrements to zero.

Default setting: 14400

Resync From SIP

Enables a resync to be triggered via a SIP NOTIFY message.

Default setting: yes

Resync After Upgrade Attempt

Triggers a resync after every firmware upgrade attempt.

Default setting: yes

Resync Trigger 1

Resync Trigger 2

Configurable resync trigger conditions. A resync is triggered when the logic equation in these parameters evaluates to TRUE.

Default setting: blank

Resync Fails On FNF

Determines whether a file-not-found response from the provisioning server constitutes a successful or a failed resync. A failed resync activates the error resync timer.

Default setting: yes

Profile Rule

This parameter is a profile script that evaluates to the provisioning resync command. The command is a TCP/IP operation and an associated URL. The TCP/IP operation can be TFTP, HTTP, or HTTPS.

If the command is not specified, TFTP is assumed, and the address of the TFTP server is obtained through DHCP option 66. In the URL, either the IP address or the FQDN of the server can be specified. The file name can have macros, such as $MA, which expands to the ATA MAC address.

Default setting: /spa$PSN.cfg

Profile Rule B:

Profile Rule C:

Profile Rule D

Defines second, third, and fourth resync commands and associated profile URLs. These profile scripts are executed sequentially after the primary Profile Rule resync operation has completed. If a resync is triggered and Profile Rule is blank, Profile Rule B, C, and D are still evaluated and executed.

Default setting: blank

DHCP Option To Use

DHCP Options, delimited by commas, retrieves firmware and profiles.

Default setting: 66.160.159.150

Transport Protocol

Transport Protocol retrieves firmware and profiles. If none is selected, TFTP is assumed and the IP address of the TFTP server is obtained from the DHCP server.

Default setting: https

Log Resync Request Msg

This parameter contains the message that is sent to the Syslog server at the start of a resync attempt.

Default setting: $PN $MAC -- Requesting resync $SCHEME://$SERVIP:$PORT$PATH

Log Resync Success Msg

Syslog message issued upon successful completion of a resync attempt.

Default setting: $PN $MAC -- Successful resync $SCHEME://$SERVIP:$PORT$PATH

Log Resync Failure Msg

Syslog message issued after a failed resync attempt.

Default setting: $PN $MAC -- Resync failed: $ERR

Report Rule

The target URL to which configuration reports are sent. This parameter has the same syntax as the Profile_Rule parameter, and resolves to a TCP/IP command with an associated URL.

A configuration report is generated in response to an authenticated SIP NOTIFY message, with Event: report. The report is an XML file containing the name and value of all the device parameters.

This parameter may optionally contain an encryption key. For example:

[ --key $K ] tftp://ps.callhome.net/$MA/rep.xml.enc

Default setting: blank

Firmware Upgrade

Field

Description

Upgrade Enable

Determines whether or not firmware upgrade operations can occur independently of resync actions.

Default setting: yes

Upgrade Error Retry Delay

The upgrade retry interval (in seconds) applied in case of upgrade failure. The ATA has a firmware upgrade error timer that activates after a failed firmware upgrade attempt. The timer is initialized with the value in this parameter. The next firmware upgrade attempt occurs when this timer counts down to zero.

Default setting: 3600

Downgrade Rev Limit

Enforces a lower limit on the acceptable version number during a firmware upgrade or downgrade. The ATA does not complete a firmware upgrade operation unless the firmware version is greater than or equal to this parameter.

Default setting: blank

Upgrade Rule

This parameter is a firmware upgrade script with the same syntax as Profile_Rule. Defines upgrade conditions and associated firmware URLs.

Default setting: blank

Log Upgrade Request Msg

Syslog message issued at the start of a firmware upgrade attempt.

Default setting: $PN $MAC -- Requesting upgrade $SCHEME://$SERVIP:$PORT$PATH

Log Upgrade Success Msg

Syslog message issued after a firmware upgrade attempt completes successfully.

Default setting: $PN $MAC -- Successful upgrade $SCHEME://$SERVIP:$PORT$PATH -- $ERR

Log Upgrade Failure Msg

Syslog message issued after a failed firmware upgrade attempt.

Default setting: $PN $MAC -- Upgrade failed: $ERR

License Keys

This field is not currently used.

CA Settings

Field

Description

Custom CA URL

The URL of a file location for a custom Certificate Authority (CA) certificate. Either the IP address or the FQDN of the server can be specified. The file name can have macros, such as $MA, which expands to the ATA MAC address.

Default setting: blank

General Purpose Parameters

Field

Description

GPP A to GPP P

General purpose provisioning parameters. These parameters can be used as variables in provisioning and upgrade rules. They are referenced by prepending the variable name with a ‘$’ character, such as $GPP_A.

Default setting: blank

Regional

Use the Voice > Regional page to localize your system with the appropriate regional settings.

Enter the settings as described below. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

Defining Ring and Cadence and Tone Scripts

To define ring and tone patterns, the ATA uses the concept of scripts. Below is information about creating Cadence Scripts (CadScripts), Frequency Scripts (FreqScripts), and Tone Scripts (ToneScripts).

CadScript

A mini-script of up to 127 characters that specifies the cadence parameters of a signal.

Syntax: S1[;S2], where:

Si=Di(oni,1/offi,1[,oni,2/offi,2[,oni,3/offi,3[,oni,4/offi,4[,oni,5/offi,5,oni,6/offi,6]]]]]) and is known as a section, oni,j and offi,j are the on/off duration in seconds of a segment and i = 1 or 2, and j = 1 to 6. Di is the total duration of the section in seconds. All durations can have up to three decimal places to provide 1 ms resolution. The wildcard character “*” represents infinite duration. The segments within a section are played in order and repeated until the total duration is played.

Example 1: 60(2/4)

Number of Cadence Sections = 1

Cadence Section 1: Section Length = 60 s

Number of Segments = 1

Segment 1: On=2s, Off=4s

Total Ring Length = 60s

Example 2—Distinctive ring (short,short,short,long): 60(.2/.2,.2/.2,.2/.2,1/4)

Number of Cadence Sections = 1

Cadence Section 1: Section Length = 60s

Number of Segments = 4

Segment 1: On=0.2s, Off=0.2s

Segment 2: On=0.2s, Off=0.2s

Segment 3: On=0.2s, Off=0.2s

Segment 4: On=1.0s, Off=4.0s

Total Ring Length = 60s

FreqScript

A mini-script of up to 127 characters that specifics the frequency and level parameters of a tone.

Syntax: F1@L1[,F2@L2[,F3@L3[,F4@L4[,F5@L5[,F6@L6]]]]

Where F1–F6 are frequency in Hz (unsigned integers only) and L1–L6 are corresponding levels in dBm (with up to 1 decimal places) White spaces before and after the comma are allowed (but not recommended)

Example 1—Call Waiting Tone: 440@-10

Number of Frequencies = 1

Frequency 1 = 440 Hz at –10 dBm

Example 2—Dial Tone: 350@-19,440@-19

Number of Frequencies = 2

Frequency 1 = 350 Hz at –19 dBm

Frequency 2 = 440 Hz at –19 dBm

ToneScript

A mini-script of up to 127 characters that specifies the frequency, level and cadence parameters of a call progress tone. May contain up to 127 characters.

Syntax: FreqScript;Z1[;Z2].

The section Z1 is similar to the S1 section in a CadScript except that each on/off segment is followed by a frequency components parameter: Z1 = D1(oni,1/offi,1/fi,1[,oni,2/offi,2/fi,2 [,oni,3/offi,3/fi,3[,oni,4/offi,4/fi,4[,oni,5/offi,5/fi,5[,oni,6/offi,6/fi,6]]]]]), where fi,j = n1[+n2]+n3[+n4[+n5[+n6]]]]] and 1 < nk < 6 indicates which of the frequency components given in the FreqScript are used in that segment; if more than one frequency component is used in a segment, the components are summed together.

Example 1—Dial tone: 350@-19,440@-19;10(*/0/1+2)

Number of Frequencies = 2

Frequency 1 = 350 Hz at –19 dBm

Frequency 2 = 440 Hz at –19 dBm

Number of Cadence Sections = 1

Cadence Section 1: Section Length = 10 s

Number of Segments = 1

Segment 1: On=forever, with Frequencies 1 and 2

Total Tone Length = 10s

Example 2—Stutter tone: 350@-19,440@-19;2(.1/.1/1+2);10(*/0/1+2)

Number of Frequencies = 2

Frequency 1 = 350 Hz at –19 dBm

Frequency 2 = 440 Hz at –19 dBm

Number of Cadence Sections = 2

Cadence Section 1: Section Length = 2s

Number of Segments = 1

Segment 1: On=0.1s, Off=0.1s with Frequencies 1 and 2

Cadence Section 2: Section Length = 10s

Number of Segments = 1

Segment 1: On=forever, with Frequencies 1 and 2

Total Tone Length = 12s

Enter the settings as described below. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

Call ProgressTones

Field

Description

Dial Tone

Prompts the user to enter a phone number. Reorder Tone is played automatically when Dial Tone or any of its alternatives times out.

Default setting: 350@-19,440@-19;10(*/0/1+2)

Second Dial Tone

Alternative to the Dial Tone when the user dials a three-way call.

Default setting: 420@-19,520@-19;10(*/0/1+2)

Outside Dial Tone

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.

Default setting: 420@-16;10(*/0/1)

Prompt Tone

Prompts the user to enter a call forwarding phone number.

Default setting: 520@-19,620@-19;10(*/0/1+2)

Busy Tone

Played when a 486 RSC is received for an outbound call.

Default setting: 480@-19,620@-19;10(.5/.5/1+2)

Reorder Tone

Played when an outbound call has failed, or after the far end hangs up during an established call. Reorder Tone is played automatically when Dial Tone or any of its alternatives times out.

Default setting: 480@-19,620@-19;10(.25/.25/1+2)

Off Hook Warning Tone

Played when the caller has not properly placed the handset on the cradle. Off Hook Warning Tone is played when the Reorder Tone times out.

Default setting: 480@-10,620@0;10(.125/.125/1+2)

Ring Back Tone

Played during an outbound call when the far end is ringing.

Default setting: 440@-19,480@-19;*(2/4/1+2)

Ring Back 2 Tone

Your ATA plays this ringback tone instead of Ring Back Tone if the called party replies with a SIP 182 response without SDP to its outbound INVITE request.

Default setting: the same as Ring Back Tone, except the cadence is 1s on and 1s off.

Default setting: 440@-19,480@-19;*(1/1/1+2)

Confirm Tone

Brief tone to notify the user that the last input value has been accepted.

Default setting: 600@-16;1(.25/.25/1)

SIT1 Tone

Alternative to the Reorder Tone played when an error occurs as a caller makes an outbound call. The RSC to trigger this tone is configurable on the SIP screen.

Default setting: 985@-16,1428@-16,1777@-16;20(.380/0/1,.380/0/2,.380/0/3,0/4/0)

SIT2 Tone

Alternative to the Reorder Tone played when an error occurs as a caller makes an outbound call. The RSC to trigger this tone is configurable on the SIP screen.

Default setting: 914@-16,1371@-16,1777@-16;20(.274/0/1,.274/0/2,.380/0/3,0/4/0)

SIT3 Tone

Alternative to the Reorder Tone played when an error occurs as a caller makes an outbound call. The RSC to trigger this tone is configurable on the SIP screen.

Default setting: 914@-16,1371@-16,1777@-16;20(.380/0/1,.380/0/2,.380/0/3,0/4/0)

SIT4 Tone

Alternative to the Reorder Tone played when an error occurs as a caller makes an outbound call. The RSC to trigger this tone is configurable on the SIP screen.

Default setting: 985@-16,1371@-16,1777@-16;20(.380/0/1,.274/0/2,.380/0/3,0/4/0)

MWI Dial Tone

Played instead of the Dial Tone when there are unheard messages in the caller’s mailbox.

Default setting: 350@-19,440@-19;2(.1/.1/1+2);10(*/0/1+2)

Cfwd Dial Tone

Played when all calls are forwarded.

Default setting: 350@-19,440@-19;2(.2/.2/1+2);10(*/0/1+2)

Holding Tone

Informs the local caller that the far end has placed the call on hold.

Default setting: 600@-19;*(.1/.1/1,.1/.1/1,.1/9.5/1)

Conference Tone

Played to all parties when a three-way conference call is in progress.

Default setting: 350@-19;20(.1/.1/1,.1/9.7/1)

Secure Call Indication Tone

Played when a call has been successfully switched to secure mode. It should be played only for a short while (less than 30 seconds) and at a reduced level (less than -19 dBm) so it does not interfere with the conversation.

Default setting: 397@-19,507@-19;15(0/2/0,.2/.1/1,.1/2.1/2)

Feature Invocation Tone

Played when a feature is implemented.

Default setting: 350@-16;*(.1/.1/1)

Call Remind Tone

The holding tone is played on Phone ports to remind user of the presence of the held call during the active call when there are two calls.

Default setting: blank

Distinctive Ring Patterns

Field

Description

Ring1 Cadence

Cadence script for distinctive ring 1.

Default setting: 60(2/4)

Ring2 Cadence

Cadence script for distinctive ring 2.

Default setting: 60(.8/.4,.8/4)

Ring3 Cadence

Cadence script for distinctive ring 3.

Default setting: 60(.4/.2,.4/.2,.8/4)

Ring4 Cadence

Cadence script for distinctive ring 4.

Default setting: 60(.3/.2,1/.2,.3/4)

Ring5 Cadence

Cadence script for distinctive ring 5.

Default setting: 1(.5/.5)

Ring6 Cadence

Cadence script for distinctive ring 6.

Default setting: 60(.2/.4,.2/.4,.2/4)

Ring7 Cadence

Cadence script for distinctive ring 7.

Default setting: 60(.4/.2,.4/.2,.4/4)

Ring8 Cadence

Cadence script for distinctive ring 8.

Default setting: 60(0.25/9.75)

Distinctive Call Waiting Tone Patterns

Field

Description

CWT1 Cadence

Cadence script for distinctive CWT 1.

Default setting: *(.3/9.7)

CWT2 Cadence

Cadence script for distinctive CWT 2.

Default setting: 30(.1/.1, .1/9.7)

CWT3 Cadence

Cadence script for distinctive CWT 3.

Default setting: 30(.1/.1, .1/.1, .1/9.7)

CWT4 Cadence

Cadence script for distinctive CWT 4.

Default setting: 30(.1/.1, .3/.1, .1/9.3)

CWT5 Cadence

Cadence script for distinctive CWT 5.

Default setting: 1(.5/.5)

CWT6 Cadence

Cadence script for distinctive CWT 6.

Default setting: 30(.1/.1,.3/.2,.3/9.1)

CWT7 Cadence

Cadence script for distinctive CWT 7.

Default setting: 30(.3/.1,.3/.1,.1/9.1)

CWT8 Cadence

Cadence script for distinctive CWT 8.

Default setting: 2.3(.3/2)

Distinctive Ring/CWT Pattern Names

Field

Description

Ring1 Name

Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 1 for the inbound call.

Default setting: Bellcore-r1

Ring2 Name

Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 2 for the inbound call.

Default setting: Bellcore-r2

Ring3 Name

Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 3 for the inbound call.

Default setting: Bellcore-r3

Ring4 Name

Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 4 for the inbound call.

Default setting: Bellcore-r4

Ring5 Name

Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 5 for the inbound call.

Default setting: Bellcore-r5

Ring6 Name

Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 6 for the inbound call.

Default setting: Bellcore-r6

Ring7 Name

Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 7 for the inbound call.

Default setting: Bellcore-r7

Ring8 Name

Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 8 for the inbound call.

Default setting: Bellcore-r8

Ring and Call Waiting Tone Spec

IMPORTANT: Ring and Call Waiting tones do not work the same way on all phones. When setting ring tones, consider the following recommendations:

- Begin with the default Ring Waveform, Ring Frequency, and Ring Voltage.

- If your ring cadence doesn’t sound right, or your phone doesn’t ring, change the following settings:

- Ring Waveform: Sinusoid

- Ring Frequency: 25

- Ring Voltage: 80

Field

Description

Ring Waveform

Waveform for the ringing signal. Choices are Sinusoid or Trapezoid.

Default setting: Trapezoid

Ring Frequency

Frequency of the ringing signal. Valid values are 15–50 (Hz)

Default setting: 20

Ring Voltage

Ringing voltage. Choices are 30–90 (V)

Default setting: 85

CWT Frequency

Frequency script of the call waiting tone. All distinctive CWTs are based on this tone.

Default setting: 440@-10

Synchronized Ring

If this is set to yes, when the ATA is called, all lines ring at the same time (similar to a regular PSTN line) After one line answers, the others stop ringing.

Default setting: no

Control Timer Values (sec)

Field

Description

Hook Flash Timer Min

Minimum on-hook time before off-hook qualifies as hook flash. Less than this the on-hook event is ignored. Range: 0.1–0.4 seconds.

Default setting: 0.1

Hook Flash Timer Max

Maximum on-hook time before off-hook qualifies as hook flash. More than this the on-hook event is treated as on hook (no hook-flash event) Range: 0.4–1.6 seconds.

Default setting: 0.9

Callee On Hook Delay

Phone must be on-hook for at this time in sec. before the ATA will tear down the current inbound call. It does not apply to outbound calls. Range: 0–255 seconds.

Default setting: 0

Reorder Delay

Delay after far end hangs up before reorder tone is played. 0 = plays immediately, inf = never plays. Range: 0–255 seconds.

Default setting: 5.

Call Back Expires

Expiration time in seconds of a call back activation. Range: 0–65535 seconds.

Default setting: 1800

Call Back Retry Intvl

Call back retry interval in seconds. Range: 0–255 seconds.

Default setting: 30

Call Back Delay

Delay after receiving the first SIP 18x response before declaring the remote end is ringing. If a busy response is received during this time, the ATA still considers the call as failed and keeps on retrying.

Default setting: 0.5

VMWI Refresh Intvl

Interval between VMWI refresh to the device.

Default setting: 0

Interdigit Long Timer

Long timeout between entering digits when dialing. The interdigit timer values are used as defaults when dialing. The Interdigit_Long_Timer is used after any one digit, if all valid matching sequences in the dial plan are incomplete as dialed. Range: 0–64 seconds.

Default setting: 10

Interdigit Short Timer

Short timeout between entering digits when dialing. The Interdigit_Short_Timer is used after any one digit, if at least one matching sequence is complete as dialed, but more dialed digits would match other as yet incomplete sequences. Range: 0–64 seconds.

Default setting: 3

CPC Delay

Delay in seconds after caller hangs up when the ATA starts removing the tip-and-ring voltage to the attached equipment of the called party. The range is 0–255 seconds. This feature is generally used for answer supervision on the caller side to signal to the attached equipment when the call has been connected (remote end has answered) or disconnected (remote end has hung up) This feature should be disabled for the called party (in other words, by using the same polarity for connected and idle state) 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.

Default setting: 2

CPC Duration

Duration in seconds for which the tip-to-ring voltage is removed after the caller hangs up. After that, tip-to-ring voltage is restored and the dial tone applies if the attached equipment is still off-hook. CPC is disabled if this value is set to 0. Range: 0 to 1.000 second. Resolution is 0.001 second.

Default setting: 0.5

Vertical Service Activation Codes

Field

Description

Call Return Code

Call Return Code This code calls the last caller.

Default setting: *69

Call Redial Code

Redials the last number called.

Default setting: *07

Blind Transfer Code

Begins a blind transfer of the current call to the extension specified after the activation code.

Default setting: *98

Call Back Act Code

Starts a callback when the last outbound call is not busy.

Default setting: *66

Call Back Deact Code

Cancels a callback.

Default setting: *86

Call Back Busy Act Code

Starts a callback when the last outbound call is busy.

Default setting: *05

Cfwd All Act Code

Forwards all calls to the extension specified after the activation code.

Default setting: *72

Cfwd All Deact Code

Cancels call forwarding of all calls.

Default setting: *73

Cfwd Busy Act Code

Forwards busy calls to the extension specified after the activation code.

Default setting: *90

Cfwd Busy Deact Code

Cancels call forwarding of busy calls.

Default setting: *91

Cfwd No Ans Act Code

Forwards no-answer calls to the extension specified after the activation code.

Default setting: *92

Cfwd No Ans Deact Code

Cancels call forwarding of no-answer calls.

Default setting: *93

Cfwd Last Act Code

Forwards the last inbound or outbound call to the number that the user specifies after entering the activation code.

Default setting: *63

Cfwd Last Deact Code

Cancels call forwarding of the last inbound or outbound call.

Default setting: *83

Block Last Act Code

Blocks the last inbound call.

Default setting: *60

Block Last Deact Code

Cancels blocking of the last inbound call.

Default setting: *80

Accept Last Act Code

Accepts the last outbound call. It lets the call ring through when do not disturb or call forwarding of all calls are enabled.

Default setting: *64

Accept Last Deact Code

Cancels the code to accept the last outbound call.

Default setting: *84

CW Act Code

Enables call waiting on all calls.

Default setting: *56

CW Deact Code

Disables call waiting on all calls.

Default setting: *57

CW Per Call Act Code

Enables call waiting for the next call.

Default setting: *71

CW Per Call Deact Code

Disables call waiting for the next call.

Default setting: *70

Block CID Act Code

Blocks caller ID on all outbound calls.

Default setting: *67

Block CID Deact Code

Removes caller ID blocking on all outbound calls.

Default setting: *68

Block CID Per Call Act Code

Blocks caller ID on the next outbound call.

Default setting: *81

Block CID Per Call Deact Code

Removes caller ID blocking on the next inbound call.

Default setting: *82

Block ANC Act Code

Blocks all anonymous calls.

Default setting: *77

Block ANC Deact Code

Removes blocking of all anonymous calls.

Default setting: *87

DND Act Code

Enables the do not disturb feature.

Default setting: *78

DND Deact Code

Disables the do not disturb feature.

Default setting: *79

CID Act Code

Enables caller ID generation.

Default setting: *65

CID Deact Code

Disables caller ID generation.

Default setting: *85

CWCID Act Code

Enables call waiting, caller ID generation.

Default setting: *25

CWCID Deact Code

Disables call waiting, caller ID generation.

Default setting: *45

Dist Ring Act Code

Enables the distinctive ringing feature.

Default setting: *26

Dist Ring Deact Code

Disables the distinctive ringing feature.

Default setting: *46

Speed Dial Act Code

Assigns a speed dial number.

Default setting: *74

Paging Code

Used for paging other clients in the group.

Default setting: *96

Secure All Call Act Code

Makes all outbound calls secure.

Default setting: *16

Secure No Call Act Code

Makes all outbound calls not secure.

Default setting: *17

Secure One Call Act Code

Makes the next outbound call secure. (It is redundant if all outbound calls are secure by default.)

Default setting: *18

Secure One Call Deact Code

Makes the next outbound call not secure. (It is redundant if all outbound calls are not secure by default.)

Default setting: *19

Conference Act Code

If this code is specified, the user must enter it before dialing the third party for a conference call. Enter the code for a conference call.

Default setting: blank

Attn-Xfer Act Code

If the code is specified, the user must enter it before dialing the third party for a call transfer. Enter the code for a call transfer.

Default setting: blank

Modem Line Toggle Code

Toggles the line to a modem. Modem passthrough mode can be triggered only by pre-dialing this code.

Default setting: *99

FAX Line Toggle Code

Toggles the line to a fax machine.

Default setting: #99

Media Loopback Code

Use for media loopback.

Default setting: *03

Referral Services Codes

These codes tell the ATA what to do when the user places the current call on hold and is listening to the second dial tone.

One or more *codes can be configured into this parameter, such as *98, or *97|*98|*123, etc. The maximum length is 79 characters. This parameter applies when the user places the current call on hold by pressing the hook flash button. Each *code (and the following valid target number according to current dial plan) triggers the ATA to perform a blind transfer to a target number that is prepended by the service *code.

For example, after the user dials *98, the ATA plays a special dial tone called the Prompt Tone while waiting for the user the enter a target number (which is checked according to dial plan as in normal dialing). When a complete number is entered, the ATA sends a blind REFER to the holding party with the Refer-To target equal to *98 target_number. This feature allows the ATA to hand off a call to an application server to perform further processing, such as call park.

The *codes should not conflict with any of the other vertical service codes internally processed by the ATA. You can empty the corresponding *code that you do not want the ATA to process.

Default setting: blank

Feature Dial Services Codes

These codes tell the ATA what to do when the user is listening to the first or second dial tone.

One or more *codes can be configured into this parameter, such as *72, or *72|*74|*67|*82, etc. The maximum length is 79 characters. This parameter applies when the user has a dial tone (first or second dial tone) After receiving dial tone, a user enters the *code and the target number according to current dial plan. For example, after user dials *72, the ATA plays a special tone called a Prompt tone while awaiting the user to enter a valid target number. When a complete number is entered, the ATA sends a INVITE to *72 target_number as in a normal call. This feature allows the proxy to process features like call forward (*72) or Block Caller ID (*67)

The *codes should not conflict with any of the other vertical service codes internally processed by the ATA. You can remove a corresponding *code that you do not want to the ATA to process.

You can add a parameter to indicate which tone plays after the *code is entered, such as *72‘c‘|*67‘p‘. Below is a list of allowed tone parameters (note the use of open quotes surrounding the parameter, without spaces)

‘c‘ = <Cfwd Dial Tone>

‘d‘ = <Dial Tone>

‘m‘ = <MWI Dial Tone>

‘o‘ = <Outside Dial Tone>

‘p‘ = <Prompt Dial Tone>

‘s‘ = <Second Dial Tone>

‘x‘ = No tones are place, x is any digit not used above

If no tone parameter is specified, the ATA plays Prompt tone by default. If the *code is not to be followed by a phone number, such as *73 to cancel call forwarding, do not include this parameter. Instead, add the *code in the dial plan and the ATA send INVITE *73@..... as usual when user dials *73.

Default setting: blank

Vertical Service Announcement Codes

Field

Description

Service Annc Base Number

Base number for service announcements.

Default setting: blank

Service Annc Extension Codes

Extension codes for service announcements.

Default setting: blank

Outbound Call Codec Selection Codes

Field

Description

Prefer G711u Code

Dial prefix to make G.711u the preferred codec for the call.

Default setting: *017110

Force G711u Code

Dial prefix to make G.711u the only codec that can be used for the call.

Default setting: *027110

Prefer G711a Code

Dial prefix to make G.711a the preferred codec for the call.

Default setting: *017111

Force G711a Code

Dial prefix to make G.711a the only codec that can be used for the call.

Default setting: *027111

Prefer G726r32 Code

Dial prefix to make G.726r32 the preferred codec for the call.

Default setting: *0172632

Force G726r32 Code

Dial prefix to make G.726r32 the only codec that can be used for the call.

Default setting: *0272632

Prefer G729a Code

Dial prefix to make G.729a the preferred codec for the call.

Default setting: *01729

Force G729a Code

Dial prefix to make G.729a the only codec that can be used for the call.

Default setting: *02729

Miscellaneous Settings

Field

Description

FXS Port Impedance:

Sets the electrical impedance of the PHONE port.

Choices are:

- 600

- 900

- 600+2.16μF

- 900+2.16μF

- 220+850||120nF

- 220+820||115nF

- 200+600||100nF

Default setting: 600.

FXS Port Input Gain:

Input gain in dB, up to three decimal places. The range is 6.000 to -12.000.

Default setting: -3.

FXS Port Output Gain:

Output gain in dB, up to three decimal places. The range is 6.000 to -12.000. The Call Progress Tones and DTMF playback level are not affected by the FXS Port Output Gain parameter.

Default setting: -3.

DTMF Playback Level:

Local DTMF playback level in dBm, up to one decimal place.

Default setting: -16.0.

DTMF Twist:

To gain difference between the two tone frequency.

Default setting: 2

DTMF Playback Length:

Local DTMF playback duration in milliseconds.

Default setting: .1.

Detect ABCD:

To enable local detection of DTMF ABCD, select Yes. Otherwise, select No. Default setting: Yes

This setting has no effect if DTMF Tx Method is INFO; ABCD is always sent OOB regardless in this setting.

Playback ABCD:

To enable local playback of OOB DTMF ABCD, select Yes. Otherwise, select No. Default setting: Yes

Caller ID Method:

Your choices are:

- Bellcore (N.Amer,China): CID, CIDCW, and VMWI. FSK sent after first ring (same as ETSI FSK sent after first ring) (no polarity reversal or DTAS).

- DTMF (Finland, Sweden): CID only. DTMF sent after polarity reversal (and no DTAS) and before first ring.

- DTMF (Denmark): CID only. DTMF sent before first ring with no polarity reversal and no DTAS.

- ETSI DTMF: CID only. DTMF sent after DTAS (and no polarity reversal) and before first ring.

- ETSI DTMF With PR: CID only. DTMF sent after polarity reversal and DTAS and before first ring.

- ETSI DTMF After Ring: CID only. DTMF sent after first ring (no polarity reversal or DTAS).

- ETSI FSK: CID, CIDCW, and VMWI. FSK sent after DTAS (but no polarity reversal) and before first ring. Waits for ACK from a device after DTAS for CIDCW.

- ETSI FSK With PR (UK): CID, CIDCW, and VMWI. FSK is sent after polarity reversal and DTAS and before first ring. Waits for ACK from a device after DTAS for CIDCW. Polarity reversal is applied only if equipment is on hook.

- DTMF (Denmark) with PR: CID only. DTMF sent after polarity reversal (and no DTAS) and before first ring.

Default setting: Bellcore(N.Amer, China)

FXS Port Power Limit:

The choices are from 1 to 8. Default setting: 3

Caller ID FSK Standard:

The ATA supports bell 202 and v.23 standards for caller ID generation. Default setting: bell 202

Feature Invocation Method:

Select the method you want to use, Default, or Sweden default. Default setting: Default.

Line 1 and Line 2 Settings (PHONE 1 and PHONE 2)

Use the Voice > Line 1 and Voice > Line 2 pages to configure the settings for calls through the PHONE 1 and PHONE 2 ports.

Enter the settings as described below. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

NOTE In a configuration profile, the FXS parameters must be appended with the appropriate numeral (for example, [1] or [2]) to identify the port to which the setting applies.

General

Field

Description

Line Enable

To enable this line for service, select yes. Otherwise, select no.

Default setting: yes

Streaming Audio Server (SAS)

Field

Description

SAS Enable

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 incoming calls and streams audio RTP packets to the caller.

Default setting: no

SAS DLG Refresh Intvl

If this value is not zero, it is the interval at which the streaming audio server sends out session refresh (SIP re-INVITE) messages to determine whether the connection to the caller is still active. If the caller does not respond to the refresh message, the ATA ends this call with a SIP BYE message. The range is 0 to 255 seconds (0 means that the session refresh is disabled)

Default setting: 30

SAS Inbound RTP Sink

The purpose of this parameter is to work around devices that do not play inbound RTP if the SAS line declares itself as a send-only device and tells the client not to stream out audio. This parameter is an FQDN or IP address of an RTP sink to be used by the SAS line in the SDP of its 200 response to inbound INVITE from a client. It will appear in the c = line and the port number, if specified, will appear in the m = line of the SDP. If this value is not specified or is equal to 0, then c = 0.0.0.0 and a=sendonly will be used in the SDP to tell the SAS client to not to send any RTP to this SAS line. If a non-zero value is specified, then a=sendrecv and the SAS client will stream audio to the given address. Special case: If the value is $IP, then the SAS line’s own IP address is used in the c = line and a=sendrecv. In that case the SAS client will stream RTP packets to the SAS line.

Default setting: blank

NAT Settings

Field

Description

NAT Mapping Enable

To use externally mapped IP addresses and SIP/RTP ports in SIP messages, select yes. Otherwise, select no.

Default setting: no

NAT Keep Alive Enable

To send the configured NAT keep alive message periodically, select yes. Otherwise, select no.

Default setting: no

NAT Keep Alive Msg

Enter the keep alive message that should be sent periodically to maintain the current NAT mapping. If the value is $NOTIFY, a NOTIFY message is sent. If the value is $REGISTER, a REGISTER message without contact is sent.

Default setting: $NOTIFY

NAT Keep Alive Dest

Destination that should receive NAT keep alive messages. If the value is $PROXY, the messages are sent to the current proxy server or outbound proxy server.

Default setting: $PROXY

Network Settings

Field

Description

SIP ToS/DiffServ Value

TOS/DiffServ field value in UDP IP packets carrying a SIP message.

Default setting: 0x68

SIP CoS Value [0-7]

CoS value for SIP messages. Valid values are 0 through 7.

Default setting: 3

RTP ToS/DiffServ Value

ToS/DiffServ field value in UDP IP packets carrying RTP data.

Default setting: 0xb8

RTP CoS Value [0- 7]

CoS value for RTP data. Valid values are 0 through 7.

Default setting: 6

Network Jitter Level

Determines how jitter buffer size is adjusted by the ATA. Jitter buffer size is adjusted dynamically. The minimum jitter buffer size is 30 milliseconds or (10 milliseconds + current RTP frame size), whichever is larger, for all jitter level settings. However, the starting jitter buffer size value is larger for higher jitter levels. This setting controls the rate at which the jitter buffer size is adjusted to reach the minimum. Select the appropriate setting: low, medium, high, very high, or extremely high.

Default setting: high

Jitter Buffer Adjustment

Choose yes to enable or no to disable this feature.

Default setting: yes

SIP Settings

Field

Description

SIP Transport

The TCP choice provides “guaranteed delivery”, which assures that lost packets are retransmitted. TCP also guarantees that the SIP packages are received in the same order that they were sent. As a result, TCP overcomes the main disadvantages of UDP. In addition, for security reasons, most corporate firewalls block UDP ports. With TCP, 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. Options are: UDP, TCP, TLS. Default setting: UDP

SIP Port

Port number of the SIP message listening and transmission port.

Default setting: 5060 for PHONE1 and 5061 for PHONE2

SIP 100REL Enable

To enable the support of 100REL SIP extension for reliable transmission of provisional responses (18x) and use of PRACK requests, select yes. Otherwise, select no.

Default setting: no

EXT SIP Port

The external SIP port number.

Default setting: blank

Auth Resync-Reboot

If this feature is enabled, the ATA authenticates the sender when it receives the NOTIFY resync reboot (RFC 2617) message. To use this feature, select yes. Otherwise, select no.

Default setting: yes

SIP Proxy-Require

The SIP proxy can support a specific extension or behavior when it sees this header from the user agent. If this field is configured and the proxy does not support it, it responds with the message, unsupported. Enter the appropriate header in the field provided.

Default setting: blank

SIP Remote-Party-ID

To use the Remote-Party-ID header instead of the From header, select yes. Otherwise, select no.

Default setting: yes

SIP GUID

This feature limits the registration of SIP accounts. The Global Unique ID is generated for each line for each ATA. When it is enabled, the ATA 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.

Default setting: no

SIP Debug Option

SIP messages are received at or sent from the proxy listen port. This feature controls which SIP messages to log. The choices are described below. Default setting: none

- none— No logging.

- 1-line— Logs the start-line only for all messages.

- 1-line excl. OPT— Logs the start-line only for all messages except OPTIONS requests/responses.

- 1-line excl. NTFY— Logs the start-line only for all messages except NOTIFY requests/responses.

- 1-line excl. REG— Logs the start-line only for all messages except REGISTER requests/responses.

- 1-line excl. OPT|NTFY|REG— Logs the start-line only for all messages except OPTIONS, NOTIFY, and REGISTER requests/responses.

- full— Logs all SIP messages in full text.

- full excl. OPT— Logs all SIP messages in full text except OPTIONS requests/responses.

- full excl. NTFY— Logs all SIP messages in full text except NOTIFY requests/responses.

- full excl. REG— Logs all SIP messages in full text except REGISTER requests/responses.

- full excl. OPT|NTFY|REG— Logs all SIP messages in full text except for OPTIONS, NOTIFY, and REGISTER requests/responses.

RTP Log Intvl

The interval for the RTP log.

Default setting: 0

Restrict Source IP

If configured, the ATA drops all packets sent to its SIP Ports from an untrusted IP address. A source IP address is untrusted if it does not match any of the IP addresses resolved from the configured Proxy (or Outbound Proxy if Use Outbound Proxy is yes)

Default setting: no

Referor Bye Delay

The number of seconds to wait before sending a BYE to the referrer to terminate a stale call leg after a call transfer.)

Default setting: 4

Refer Target Bye Delay

The number of seconds to wait before sending a BYE to the refer target to terminate a stale call leg after a call transfer.)

Default setting: 0

Referee Bye Delay

The number of seconds to wait before sending a BYE to the referee to terminate a stale call leg after a call transfer.)

Default setting: 0

Refer-To Target Contact

To contact the refer-to target, select yes. Otherwise, select no.

Default setting: no

Sticky 183

If this feature is enabled, the ATA ignores further 180 SIP responses after receiving the first 183 SIP response for an outbound INVITE. To enable this feature, select yes. Otherwise, select no.

Default setting: no

Auth INVITE

When enabled, authorization is required for initial incoming INVITE requests from the SIP proxy.

Default setting: no

Reply 182 On Call Waiting

When enabled, the ATA replies with a SIP182 response to the caller if it is already in a call and the line is off-hook. To use this feature select yes.

Default setting: no

Use Anonymous With RPID

Determines whether or not the ATA uses “Anonymous” when Remote Party ID is requested in the SIP message.

Default setting: yes

Use Local Addr In From

Use the local ATA IP address in the SIP FROM message.

Default setting: no

Call Feature Settings

Field

Description

Blind Attn-Xfer Enable

Enables the ATA to perform an attended transfer operation by ending the current call leg and performing a blind transfer of the other call leg. If this feature is disabled, the ATA performs an attended transfer operation by referring the other call leg to the current call leg while maintaining both call legs. To use this feature, select yes. Otherwise, select no.

Default setting: no

MOH Server

User ID or URL of the auto-answering streaming audio server. When 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.

Default setting: blank

Xfer When Hangup Conf

Makes the ATA perform a transfer when a conference call has ended. Select yes or no from the drop-down menu.

Default setting: yes

Conference Bridge URL

This feature supports external conference bridging for n-way conference calls (n>2), instead of mixing audio locally. To use this feature, set this parameter to that of the server's name. For example: conf@mysefver.com:12345 or conf (which uses the Proxy value as the domain).

Default setting: blank

Conference Bridge Ports

Select the maximum number of conference call participants. The range is 3 to 10.

Default setting: 3

Enable IP Dialing

Enable or disable IP dialing. If IP dialing is enabled, one can dial [userid@] 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 and 255, and port must be larger than 255. If port is not given, 5060 is used. Port and User-Id are optional. 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.

Default setting: no

Emergency Number

Comma separated list of emergency number patterns. If outbound call matches one of the pattern, the ATA will disable hook flash event handling. The condition is restored to normal after the call ends. Blank signifies that there is no emergency number. Maximum number length is 63 characters.

Default setting: blank

Mailbox ID

Enter the ID number of the mailbox for this line.

Default setting: blank

Feature Key Sync

Allows the phone to synchronize with the call server. If yes is selected, if Do Not Disturb or Call Forwarding settings are changed on the phone, changes are also made on the server; if changes are made on the server, they are propagated to the phone.

Default setting: no

Proxy and Registration

Field

Description

Proxy

SIP proxy server for all outbound requests.

Default setting: blank

Outbound Proxy

SIP Outbound Proxy Server where all outbound requests are sent as the first hop.

Default setting: blank

Use Outbound Proxy

Enables the use of an Outbound Proxy. If set to no, the Outbound Proxy and Use OB Proxy in Dialog parameters are ignored.

Default setting: no

Use OB Proxy In Dialog

Whether to force SIP requests to be sent to the outbound proxy within a dialog. Ignored if the parameter Use Outbound Proxy is no, or the Outbound Proxy parameter is empty.

Default setting: yes

Register

Enable periodic registration with the Proxy parameter. This parameter is ignored if Proxy is not specified.

Default setting: yes

Make Call Without Reg

Allow making outbound calls without successful (dynamic) registration by the unit. If No, dial tone will not play unless registration is successful.

Default setting: no

Register Expires

Expires value in sec in a REGISTER request. The ATA will periodically renew registration shortly before the current registration expired. This parameter is ignored if the Register parameter is no. Range: 0 – (231 – 1) sec.

Default setting: 3600

Ans Call Without Reg

Allow answering inbound calls without successful (dynamic) registration by the unit.

Default setting: no

Use DNS SRV

Whether to use DNS SRV lookup for Proxy and Outbound Proxy.

Default setting: no

DNS SRV Auto Prefix

If enabled, the ATA will automatically prepend the Proxy or Outbound Proxy name with _sip._udp when performing a DNS SRV lookup on that name.

Default setting: no

Proxy Fallback Intvl

After failing over to a lower priority server, the ATA waits for the specified Proxy Fallback Interval, in seconds, before retrying the highest priority proxy (or outbound proxy) servers. This parameter is useful only if the primary and backup proxy server list is provided to the ATA via DNS SRV record lookup on the server name. (Using multiple DNS A records per server name does not allow the notion of priority, so all hosts will be considered at the same priority and the ATA will not attempt to fall back after a failover.)

Default setting: 3600

Proxy Redundancy Method

The method that the ATA uses to create a list of proxies returned in the DNS SRV records. If you select Normal, the list will contain proxies ranked by weight and priority. If you select Based on SRV port, the ATA also inspects the port number based on 1st proxy’s port.

Default setting: Normal

Mailbox Subscribe URL

The URL or IP address of the voicemail server.

Default setting: blank

Mailbox Subscribe Expires

Sets subscription interval for voicemail message waiting indication. When this time period expires, the ATA sends another subscribe message to the voice mail server. Default: 2147483647

Auto Register when Failover

This parameter controls the failover behavior when there is an error.

Default setting: no.

There are two options:

- No - Resend directly to backup proxy.

- Yes - Unregister with current proxy, register to backup proxy and then resend to backup proxy.

This option is used under the following conditions:

- The failed message is an INVITE message.

- The return code matches one of the values in parameter Try Backup RSC.

Subscriber Information

Field

Description

Display Name

Display name for caller ID.

Default setting: blank

User ID

User ID for this line.

Default setting: blank

Password

Password for this line.

Default setting: blank

Use Auth ID

To use the authentication ID and password for SIP authentication, select yes. Otherwise, select no to use the user ID and password.

Default setting: no

Auth ID

Authentication ID for SIP authentication.

Default setting: blank

Resident Online Number

This setting allows you to associate a "local" telephone number with this line using a valid Skype Online Number from Skype. Calls made to that number will ring your phone. Enter the number without spaces or special characters.

Default setting: blank

SIP URI

The parameter by which the user agent will identify itself for this line. If this field is blank, the actual URI used in the SIP signaling should be automatically formed as:

sip:UserName@Domain

Where UserName is the username given for this line in the User ID, and Domain is the domain given for this profile in the User Agent Domain. If the User Agent Domain is an empty string, then the IP address of the phone should be used for the domain.

If the URI field is not empty, but if a SIP or SIPS URI that contains no @ character, then the actual URI used in the SIP signaling should be automatically formed by appending this parameter with an @ character followed by the IP address of the device.

Supplementary Service Subscription

The ATA provides native support of a large set of enhanced or supplementary services. All of these services are optional. The parameters listed in the following table are used to enable or disable a specific supplementary service. A supplementary service should be disabled if a) 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.

Field

Description

Call Waiting Serv

Enable Call Waiting Service.

Default setting: yes

Block CID Serv

Enable Block Caller ID Service.

Default setting: yes

Block ANC Serv

Enable Block Anonymous Calls Service

Default setting: yes

Dist Ring Serv

Enable Distinctive Ringing Service

Default setting: yes

Cfwd All Serv

Enable Call Forward All Service

Default setting: yes

Cfwd Busy Serv

Enable Call Forward Busy Service

Default setting: yes

Cfwd No Ans Serv

Enable Call Forward No Answer Service

Default setting: yes

Cfwd Sel Serv

Enable Call Forward Selective Service. Configure this service in the Selective Call Forward Settings section.

Default setting: yes

Cfwd Last Serv

Enable Forward Last Call Service

Default setting: yes

Block Last Serv

Enable Block Last Call Service

Default setting: yes

Accept Last Serv

Enable Accept Last Call Service

Default setting: yes

DND Serv

Enable Do Not Disturb Service

Default setting: yes

CID–Serv

Enable Caller ID Service

Default setting: yes

CWCID Serv

Enable Call Waiting Caller ID Service

Default setting: yes

Call Return Serv

Enable Call Return Service

Default setting: yes

Call Redial Serv

Enable Call Redial Service.

Default setting: yes

Call Back Serv

Enable Call Back Service.

Default setting: yes

Three Way Call Serv

Enable Three Way Calling Service. Three Way Calling is required for Three Way Conference and Attended Transfer.

Default setting: yes

Three Way Conf Serv

Enable Three Way Conference Service. Three Way Conference is required for Attended Transfer.

Default setting: yes

Attn Transfer Serv

Enable Attended Call Transfer Service. Three Way Conference is required for Attended Transfer.

Default setting: yes

Unattn Transfer Serv

Enable Unattended (Blind) Call Transfer Service.

Default setting: yes

MWI Serv

Enable MWI Service. MWI is available only if a Voice Mail Service is set-up in the deployment.

Default setting: yes

VMWI Serv

Enable VMWI Service (FSK)

Default setting: yes

Speed Dial Serv

Enable Speed Dial Service.

Default setting: yes

Secure Call Serv

Secure Call Service. If this feature is enabled, a user can make a secure call by entering an activation code (*18 by default) before dialing the target number. Then audio traffic in both directions is encrypted for the duration of the call.

Default setting: yes

Star codes are set in Vertical Service Activation Codes. To enable secure calling by default, without requiring a star code, set the user’s Secure Call Setting to yes. See User 1 and User 2.

Referral Serv

Enable Referral Service. See the Referral Services Codes parameter For more information.

Default setting: yes

Feature Dial Serv

Enable Feature Dial Service. See the Feature Dial Services Codes parameter For more information.

Default setting: yes

Service Announcement Serv

Enable Service Announcement Service.

Default setting: no

Reuse CID Number As Name

Use the Caller ID number as the caller name.

Default settings: yes

CONFCID Serv

Enable Caller ID during conference call.

Default settings: yes

Audio Configuration

Field

Description

Preferred Codec

Preferred codec for all calls. (The actual codec used in a call still depends on the outcome of the codec negotiation protocol.) Select one of the following:

- G711u

- G711a

- G726-32

- G729a

Default setting: G711u.

Second Preferred Codec

If the first codec fails, then second preferred codec is tried.

Default setting: blank

Third Preferred Codec

If the second codec fails, then third preferred codec is tried.

Default setting: blank

To use only the preferred codec for all calls, select yes. (The call fails if the far end does not support this codec.) Otherwise, select no.

Default setting: no

Use Remote Pref Codec

If set to yes, the phone communicates using the remote phone preferred codec. If set to no, the Cisco IP phone communicates using its own preferred codec (as indicated in the Preferred Codec field and in the SDP by order of preferences).

Default setting: no

Codec Negotiation

When set to Default, the Cisco IP phone responds to an Invite with a 200 OK response advertising the preferred codec only. When set to List All, the Cisco IP phone responds listing all the codecs that the phone supports.

Default setting: Default

G729a Enable

To enable the use of the G.729a codec at 8 kbps, select yes. Otherwise, select no.

Default setting: yes

Silence Supp Enable

To enable silence suppression so that silent audio frames are not transmitted, select yes. Otherwise, select no.

Default setting: no

G726-32 Enable

To enable the use of the G.726 codec at 32 kbps, select yes. Otherwise, select no.

Default setting: yes

Silence Threshold

Select the appropriate setting for the threshold: high, medium, or low.

Default setting: medium

FAX V21 Detect Enable

To enable detection of V21 fax tones, select yes. Otherwise, select no.

Default setting: yes

Echo Canc Enable

To enable the use of the echo canceller, select yes. Otherwise, select no.

Default setting: yes

FAX CNG Detect Enable

To enable detection of the fax Calling Tone (CNG), select yes. Otherwise, select no.

Default setting: yes

FAX Passthru Codec

Select the codec for fax passthrough, G711u or G711a.

Default setting: G711u

FAX Codec Symmetric

To force the ATA to use a symmetric codec during fax passthrough, select yes. Otherwise, select no.

Default setting: yes

DTMF Process INFO

To use the DTMF process info feature, select yes. Otherwise, select no.

Default setting: yes

FAX Passthru Method

Select the fax passthrough method: None, NSE, or ReINVITE.

Default setting: NSE

DTMF Process AVT

To use the DTMF process AVT feature, select yes. Otherwise, select no.

Default setting: yes

FAX Process NSE

To use the fax process NSE feature, select yes. Otherwise, select no.

Default setting: yes

DTMF Tx Method

Select the method to transmit DTMF signals to the far end: InBand, AVT, INFO, or Auto. InBand sends DTMF by using the audio path. AVT sends DTMF as AVT events. INFO uses the SIP INFO method. Auto uses InBand or AVT based on the outcome of codec negotiation.

Default setting: Auto

FAX Disable ECAN

If enabled, this feature automatically disables the echo canceller when a fax tone is detected. To use this feature, select yes. Otherwise, select no.

Default setting: no

DTMF Tx Mode

DTMF Detection Tx Mode is available for SIP information and AVT. Options are: Strict or Normal.

Default setting: Strict for which the following are true:

- A DTMF digit requires an extra hold time after detection.

- The DTMF level threshold is raised to -20 dBm.

The minimum and maximum duration thresholds are:

- strict mode for AVT and SIP: the value set in DTMF Tx Strict Hold Off Time

- normal mode for AVT: 40 ms

- normal mode for SIP: 50 ms

DTMF Tx Strict Hold Off Time

This parameter is in effect only when DTMF Tx Mode is set to strict, and when DTMF Tx Method is not set to inband; i.e. either AVT or 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 setting: 70 ms

FAX Enable T38

To enable the use of ITU-T T.38 standard for FAX Relay, select yes. Otherwise select no.

Default setting: no

Hook Flash Tx Method

Select the method for signaling hook flash events: None, AVT, or INFO. None does not signal hook flash events. AVT uses RFC2833 AVT (event = 16) INFO uses SIP INFO with the single line signal=hf in the message body. The MIME type for this message body is taken from the Hook Flash MIME Type setting.

Default setting: None

FAX T38 Redundancy

Select the appropriate number to indicate the number of previous packet payloads to repeat with each packet. Choose 0 for no payload redundancy. The higher the number, the larger the packet size and the more bandwidth consumed.

Default setting: 1

FAX T38 ECM Enable

Select yes to enable T.38 Error Correction Mode. Otherwise select no.

Default setting: yes

FAX Tone Detect Mode

This parameter has three possible values:

- caller or callee : The ATA will detect FAX tone whether it is callee or caller

- caller only: The ATA will detect FAX tone only if it is the caller

- callee only : The ATA will detect FAX tone only if it is the callee

Default setting: caller or callee.

Symmetric RTP

Enable symmetric RTP operation. If enabled, the ATA sends RTP packets to the source address and port of the last received valid inbound RTP packet. If disabled (or before the first RTP packet arrives) the ATA sends RTP to the destination as indicated in the inbound SDP.

Default setting: no

Fax T38 Return to Voice

When this feature is enabled, upon completion of the fax image transfer, the connection remains established and reverts to a voice call using the previously designated codec. Select yes to enable this feature, or select no to disable it.

Default setting: no

Modem Line

Enable an alternate method to make the modem call without Modem Line Toggle Code pre-dialing.

Default setting: no

RTP to Proxy in Remote Hold

Enable to send RTP to proxy when line is held by remote side.

Default setting: no

The default dial plan script for the line is as follows: (*xx|[3469]11|0|00|[2- 9]xxxxxx|1xxx[2-9]xxxxxx|xxxxxxxxxxxx.)

Each parameter is separated by a semi-colon (;)

Example 1:

*1xxxxxxxxxx<:@fwdnat.pulver.com:5082;uid=jsmith;pwd=xy z

Example 2:

*1xxxxxxxxxx<:@fwd.pulver.com;nat;uid=jsmith;pwd=xyz

The syntax for a dial plan expression is described in the table below.

Dial Plan Entry

Functionality

*xx

Allow arbitrary 2 digit star code

[3469]11

Allow x11 sequences

0

Operator

00

Int’l Operator

[2-9]xxxxxx

US local number

1xxx[2-9]xxxxxx

US 1 + 10-digit long distance number

xxxxxxxxxxxx.

Everything else

FXS Port Polarity Configuration

Field

Description

Idle Polarity

Polarity before a call is connected: Forward or Reverse.

Default setting: Forward

Caller Conn Polarity

Polarity after an outbound call is connected: Forward or Reverse.

Default setting: Forward.

Callee Conn Polarity

Polarity after an inbound call is connected: Forward or Reverse.

Default setting: Forward

User 1 and User 2

Use the Voice > User 1 and Voice > User 2 pages to set the user preferences for the calls through the PHONE 1 and PHONE 2 ports.

Enter the settings as described below. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

Call Forward Settings

Field

Description

Cfwd All Dest

Forward number for Call Forward All Service.

Default setting: blank

Cfwd Busy Dest

Forward number for Call Forward Busy Service. Same as Cfwd All Dest.

Default setting: blank

Cfwd No Ans Dest

Forward number for Call Forward No Answer Service. Same as Cfwd All Dest.

Default setting: blank

Cfwd No Ans Delay

Delay in sec before Call Forward No Answer triggers.

Default setting: 20

Selective Call Forward Settings

Field

Description

Cfwd Sel1-8 Caller

Caller number pattern to trigger Call Forward Selective service. When the caller’s phone number matches the entry, the call is forwarded to the corresponding Cfwd Selective Destination (Cfwd Sel1-8 Dest).

- Use ? to match any single digit.

- Use * to match any number of digits.

Example: 1408*, 1512???1234

In the above example, a call is forwarded to the corresponding destination if the caller ID either starts with 1408 or is an 11-digit numbering starting with 1512 and ending with 1234.

Default setting: blank

Cfwd Sel1-8 Dest

The destination for the corresponding Call Forward Selective caller pattern (Cfwd Sel1-8 Caller).

Default setting: blank

Cfwd Last Caller

The number of the last caller; this caller is actively forwarded to the Cfwd Last Dest via the Call Forward Last service. For more information, see Vertical Service Activation Codes.

Default setting: blank

Cfwd Last Dest

The destination for the Cfwd Last Caller.

Block Last Caller

The number of the last caller; this caller is blocked via the Block Last Caller Service. For more information, see Vertical Service Activation Codes.

Default setting: blank

Accept Last Caller

The number of the last caller; this caller is accepted via the Accept Last Caller Service. For more information, see Vertical Service Activation Codes.

Default setting: blank

Speed Dial Settings

Field

Description

Speed Dial 2-9

Target phone number (or URL) assigned to speed dial 2, 3, 4, 5, 6, 7, 8, or 9.

Default setting: blank

Supplementary Service Settings (User)

Field

Description

CW Setting

Call Waiting on/off for all calls.

Default setting: yes

Block CID Setting

Block Caller ID on/off for all calls.

Default setting: no

Block ANC Setting

Block Anonymous Calls on or off.

Default setting: no

DND Setting

DND on or off.

Default setting: no

CID Setting

Caller ID Generation on or off.

Default setting: yes

CWCID Setting

Call Waiting Caller ID Generation on or off.

Default setting: yes

Dist Ring Setting

Distinctive Ring on or off.

Default setting: yes

Secure Call Setting

If yes, all outbound calls are secure calls by default, without requiring the user to dial a star code first.

Default setting: no

- If Secure Call Setting is set to yes, all outbound calls are secure. However, a user can disable security for a call by dialing *19 before dialing the target number.

- If Secure Call Setting is set to No, the user can make a secure outbound call by dialing *18 before dialing the target number.

- A user cannot force inbound calls to be secure or not secure; that depends on whether the caller has security enabled or not.

Note: This setting is applicable only if Secure Call Serv is set to yes on the line interface. See Line 1 and Line 2 Settings (PHONE 1 and PHONE 2).

Message Waiting

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.

Default setting: no

Accept Media Loopback Request

Controls how to handle incoming requests for loopback operation.

Default setting: automatic

- never: Never accepts loopback calls; replies 486 to the caller.

- automatic: Automatically accepts the call without ringing.

- manual: Rings the phone first, and the call must be picked up manually before loopback starts.

Default setting: Automatic

Media Loopback Mode

The loopback mode to assume locally when making call to request media loopback. Choices are: Source and Mirror.

Default setting: source

NOTE If the ATA answers the call, the mode is determined by the caller.

Media Loopback Type

The loopback type to use when making call to request media loopback operation. Choices are Media and Packet.

Default setting: media

Note that if the ATA answers the call, then the loopback type is determined by the caller (the ATA always picks the first loopback type in the offer if it contains multiple type)

CONFCID Setting

Enables or disables the CONFCID.

Default setting: yes

Distinctive Ring Settings

Field

Description

Ring1 - 8 Caller

Caller number pattern to play Distinctive Ring/CWT 1, 2, 3, 4, 5, 6, 7, or 8. Caller number patterns are matched from Ring 1 to Ring 8. The first match (not the closest match) will be used for alerting the subscriber. The distinctive rings are set on the Regional page.

Default setting: blank

Ring Settings

Field

Description

Default Ring

Default ringing pattern, 1–8, for all callers.

Default setting: 1

Default CWT

Default CWT pattern, 1–8, for all callers.

Default setting: 1

Hold Reminder Ring

Ring pattern for reminder of a holding call when the phone is on-hook.

Default setting: 8

Call Back Ring

Ring pattern for call back notification.

Default setting: 7

Cfwd Ring Splash Len

Duration of ring splash when a call is forwarded (0 – 10.0s)

Default setting: 0

Cblk Ring Splash Len

Duration of ring splash when a call is blocked (0 – 10.0s)

Default setting: 0

VMWI Ring Policy

The parameter controls when a ring splash is played when a the VM server sends a SIP NOTIFY message to the ATA indicating the status of the subscriber’s mail box. Three settings are available.

Default setting: New VM Available

- New VM Available: Ring as long as there new voicemail messages.

- New VM Becomes Available: Ring at the point when the first new voicemail message is received.

- New VM Arrives: Ring when the number of new voicemail messages increases.

VMWI Ring Splash Len

Duration of ring splash when new messages arrive before the VMWI signal is applied (0 – 10.0s)

Default setting: 0

Ring On No New VM

If enabled, the ATA plays a ring splash when the voicemail server sends SIP NOTIFY message to the ATA indicating that there are no more unread voice mails. Some equipment requires a short ring to precede the FSK signal to turn off VMWI lamp.

Default setting: no

Administration Settings

This chapter describes the administrative settings for the ATA. It includes the following sections:

- Management

- Logging

- Diagnostics

- Factory Defaults

- Firmware Upgrade

- Configuration Management

- Reboot

Management

Use the Management pages to manage web access to the configuration utility and to enable protocols for remote configuration and network management.

- Web Access Management

- TR-069

- SNMP

- User List (Password Management)

- Bonjour

- Reset Button

Web Access Management

Use the Administration > Management > Web Access Management page to configure the settings for access to the administration of the ATA.

Cisco SPA112 Web Access Management fields

Access to the Cisco SPA112 configuration utility is enabled by default. Admin Access allows you to manage the configuration from a computer in your office network, and Web Utility Access allows you to connect from a computer on a different subnet or on the Internet. To access the configuration utility, launch a web browser and enter the URL in the Address bar. The URL must include the specified protocol, the WAN IP address of the ATA, and the specified port number. For example, with the HTTPS protocol, a WAN IP address of 203.0.113.50, and port 80, you would enter: https://203.0.113.50:80

Field

Description

Admin Access

This feature controls access to the configuration utility from devices that are connected via the ETHERNET (LAN) port.

Click Enabled to enable this feature, or click Disabled to disable it. The default setting is Enabled. If you administer and configure the ATA from a computer that is connected to the LAN, this feature must be enabled.

Web Utility Access

Select the protocol to use for access to the configuration utility from a device on the WAN. Choose HTTP and/or HTTPS. For secure Internet access, select HTTPS. The default value is HTTP.

Remote Management Port

Enter the port number to use for access to the configuration utility from a device on the WAN. The default port number is 80.

Cisco SPA122 Web Access fields

Field

Description

Admin Access

This feature controls access to the configuration utility from devices that are connected via the ETHERNET (LAN) port.

Click Enabled to enable this feature, or click Disabled to disable it. The default setting is Enabled. If you administer and configure the ATA from a computer that is connected to the LAN, this feature must be enabled.

Web Utility Access

Select the protocol to use for access to the configuration utility from a device on the WAN. Choose HTTP and/or HTTPS. For secure Internet access, select HTTPS. The default value is HTTP.

Cisco SPA122 Remote Access fields

Field

Description

Remote Management

Allows access to the configuration utility from a device that is on the WAN side of the ATA. For example, you could connect from another subnet in your office or from your home computer.

Click Enabled to enable this feature, or click Disabled to disable it. The default setting is Disabled. The other fields in this section of the page are available only if you enable this feature.

If you attempt to enable this feature while using the default administrator login credentials, you will be prompted to change the credentials. Click OK to acknowledge the warning message. Use the Administration > Management > User List page to change administrator password. For more information, see User List (Password Management).

Web Utility Access

Select the protocol to use for access to the configuration utility from a device on the WAN side of the ATA. Choose HTTP and/or HTTPS. For secure Internet access, select HTTPS. The default value is HTTP.

Include the specified protocol when you enter the address in your web browser. For example, with the HTTPS protocol, a WAN IP address of 203.0.113.50, and the default Remote Management Port of 80, you would enter: https://203.0.113.50:80

Remote Upgrade

If you enabled Remote Management, choose whether or not to allow firmware upgrades from a device on the WAN side of the ATA. Click Enabled to enable this feature, or click Disabled to disable it. The default value is Disabled.

You can change this setting only when your computer is connected to the configuration utility from the LAN.

Allowed Remote IP Address

You can use this feature to limit access to the configuration utility based on the IP address of a device. Choose Any IP Address to allow access from any external IP address. To specify an external IP address or range of IP addresses, select the second radio button and then enter the desired IP address or range. The default setting is Any IP Address.

Remote Management Port

Enter the port number to use for access to the configuration utility from a device on the WAN side of the ATA. The default port number is 80.

Include the specified port when you enter the address in your web browser. For example, with the HTTPS protocol, a WAN IP address of 203.0.113.50, and the default Remote Management Port of 80, you would enter: https://203.0.113.50:80

TR-069

Use the Administration > Management > TR-069 page to configure communication with an Auto-Configuration Server (ACS) via TR-069 CPE WAN Management Protocol (CWMP). TR-069 (Technical Report 069) provides a common platform to manage all voice devices and other customer-premises equipment (CPE) in large-scale deployments. It provides the communication between the CPE and the ACS.

Enter the settings as described below. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

Field

Description

Status

Click Enabled to enable remote provisioning, or click Disabled to disable this feature. The default setting is Disabled.

ACS URL

The URL for the ACS. The format should be http(s)://xxx.xxx.xxx.xxx:port or xxx.xxx.xxx.xxx:port. The xxx.xxx.xxx.xxx is the domain name or IP address of the ACS server. Both the IP address and the port number are required.

ACS Username

The username for the ACS. The default username is the Organization Unit Identifier (OUI). This value is required and must match the username configured on the ACS.

ACS Password

The password for the ACS. This value is required and must match the password configured on the ACS.

Connection Request Port

The port to use for connection requests

Connection Request Username

The username for connection requests. This value must match the Connection Request Username configured on the ACS.

Connection Request Password

The password for connection requests. This value must match the Connection Request Password configured on the ACS.

Periodic Inform Interval

If Periodic Inform is enabled, the duration, in seconds, between CPE attempts to connect to the ACS. The default value is 86400 seconds.

Periodic Inform Enable

Click Enabled to enable CPE connection requests to the ACS, or click Disabled to disable this feature.

Request Download

If applied, ACS may call the Download RPC after it receives the request from the ATA.

SNMP

Use the Administration > Management > SNMP page to set up Simple Network Management Protocol (SNMP) for the ATA.

SNMP is a network protocol that allows network administrators to manage, monitor, and receive notifications of critical events as they occur on the network. The ATA supports SNMPv2 and SNMPv3. It acts as an SNMP agent that replies to SNMP commands from SNMP Network Management Systems. It supports the standard SNMP get, next, and set commands. It also generates SNMP traps to notify the SNMP manager when configured alarm conditions occur. Examples include reboots, power cycles, and INTERNET (WAN) events.

Enter the settings as described below. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

Settings for SNMPv2

Field

Description

Enabled, Disabled

Click Enabled to enable this feature, or click Disabled to disable it. The default setting is Disabled.

Trusted IP

Choose Any to allow access from any IP address (not recommended). Click Address to specify the IP address and subnet mask of a single SNMP manager or trap agent that can access the ATA through SNMP.

Get/Trap Community

Enter a community string for authentication for SNMP GET commands. The default value is public.

Set Community

Enter a community string for authentication for SNMP SET commands. The default value is private.

Settings for SNMPv3

Field

Description

Enabled, Disabled

Click Enabled to enable this feature, or click Disabled to disable it. The default setting is Disabled.

R/W User

Enter the user name for SNMPv3 authentication. The default value is v3rwuser.

Auth-Protocol

Choose the SNMPv3 authentication protocol from the drop-down list (HMAC-MD5 or HMAC-SHA).

Auth-Password

Enter the authentication password.

PrivProtocol

Choose a privacy authentication protocol from the drop-down list (None or CBC-DES). If you select CBCDES, the privKey encrypts the data portion of the message that is being sent.

Privacy Password

Enter the key for the authentication protocol to use.

Trap Configuration

Field

Description

IP Address

The IP Address of the SNMP manager or trap agent.

Port

The SNMP trap port used by the SNMP manager or trap agent to receive the trap messages. Valid entries are 162 or 1025~65535. The default value is162.

SNMP Version

The SNMP version in use by the SNMP manager or trap agent. Choose a version from the list.

User List (Password Management)

Use the Administration > Management > User List page to manage the two user accounts for the configuration utility. The administrator-level account has the default username admin and password admin. The user-level account has access to modify a limited set of features. This account has the default username cisco and password cisco.

For the IVR, no user password is required; the user simply presses # when prompted. The default administrator password is 1234#. You can configure these passwords on the System page.

To update a password:

STEP 1 In the User List table, click the pencil icon for the account that you want to update.

STEP 2 On the User Account page, enter the username and password, as described below.

- Username: Enter a username.

- Old Password (administrator account only): Enter the existing password. The default administrator password is admin . The default guest password is cisco .

- New Password: Enter up to 32 characters for your new password.

- Confirm New Password: Enter the new password again, to confirm.

STEP 3 After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

Bonjour

Use the Administration > Management > Bonjour page to enable or disable Bonjour. Bonjour is a service discovery protocol that locates network devices such as computers and servers on your LAN. It may be required by network management systems that you use. When this feature is enabled, the ATA periodically multicasts Bonjour service records to its entire local network to advertise its existence.

Click Enabled to enable this feature, or click Disabled to disable it. The default setting is Enabled. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

Reset Button

Click Enabled to enable the reset button, or click Disabled to disable it. The default setting is Enabled. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

Logging

The ATA allows you to record incoming, outgoing, and DHCP lists for various events that occur on your network.The Incoming Log displays a temporary list of the source IP addresses and destination port numbers for the incoming Internet traffic. The Outgoing Log displays a temporary list of the local IP addresses, destination URLs/IP addresses, and service/port numbers for the outgoing Internet traffic.

See these topics:

- Log Module

- Log Setting

- Log Viewer

Log Module

Use the Administration > Log > Log Module page to enable and configure logging.

- As a best practice, Cisco recommends that you enable logging only when needed, and disable logging when you finish the investigation. Logging consumes resources and can impact system performance.

- If you want to enable email or syslog server logging, first specify the email or syslog server settings on the Log Setting page.

Enter the settings as described below. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

Field

Description

Status

Click Enabled to enable logging, or click Disabled to disable logging. The default setting is Disabled.

Log - Enable

Check the box in the heading row to enable logging for all services (kernel and system). Alternatively, check the box for kernel or system to enable logging for that service.

Service

The type of service to include: kernel or system.

Priority

The types of events to include in the log. The lowest level of logging is Emergency, which is limited to messages about high impact events. The highest level of logging is Debugging, which includes all message types from Emergency upward.

- Emergency: Messages about events, such as an imminent system crash, that make the system unusable. Typically this type of message is broadcast to all users.

- Alert: Messages about conditions, such as a corrupted system database, that require immediate corrective action.

- Critical: Messages about serious conditions, such as a disk failure.

- Error: Messages about conditions that require corrective action but are not critical.

- Warning: Warnings about possible issues.

- Notification: Messages about normal but significant conditions that may require attention.

- Information: Messages that provide information only.

- Debugging: Messages that are used to debug programs.

Local

Check the box in the heading row to include all services in the local logs that can be viewed in the Log Viewer. Alternatively, check the box for kernel or system to include that service in the local log.

E-Mail

Check the box in the heading row to include all services in the emailed logs, if configured on the Log Setting page. Alternatively, check the box for kernel or system to include that service in the emailed log.

Syslog Server

Check the box in the heading row to include all services in the log file that is transmitted to the syslog server. Alternatively, check the box for kernel or system to include that service in the log file.

Log Setting

If logging is enabled on the Administration > Log > Log Module page, the ATA can periodically send the log file to a server or to an email address. Use the Log Setting page to enter the information for your syslog server and email account.

NOTE For information about enabling and configuring logging, see Log Module.

Enter the settings as described below. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

Local

Field

Description

Log Size

Enter the maximum size of the log file in kilobytes. Valid values are from 128 to 1024.

Syslog Server

Field

Description

IP Address

Enter the IP address of the syslog server where the messages will be sent.

Port

Enter the port to use on the server. Valid values are from 1 to 65535.

E-Mail

When logging is enabled, you can send logs to an email address by using SMTP.

NOTE Service providers’ requirements vary. Be aware that some providers do not allow SMTP email from a free account. Other providers may require a user to log on to a new mailbox before sending emails. For accurate information, read the support documentation from your provider. In your provider's support or help system, search for information about SMTP server settings.

Field

Description

Sender

If you wish to send log entries to an email account, complete all of the fields in this section. Enter a valid email address to identify the sender of the email. Example: user1@company.com

Receiver

Enter a valid email address where the email will be sent. Example: user2@company.com

SMTP Server

Enter the IP address or domain name of the mail server that you will use to send this email. Example: smtp.gmail.com

SMTP Port

Enter the port to use on the SMTP server. The default value is 25. Use the port specified by your email server administrator or service provider.

Subject

Enter a brief description for the subject line of the email. Example: Log from My ATA

Number of Logs

Enter the number of log entries to include in the email. The valid range is 10 to 200.

Interval

Enter the interval, in minutes, at which to send emails. The valid range is 1 to 1440 (24 hours).

Username

Enter the username for the email account that will be used to send these emails. Use the format required by your service provider. Usually it is the full email address. Example: user1@company.com.

Password

Enter the password for the email account that will be used to send these emails.

Log Viewer

If logging is enabled on the Administration > Log > Log Module page, you can use the Log Viewer page view the logs online and to download the system log file to your computer. You can limit the contents of the log by choosing the types of entries to include and by specifying keywords.

NOTE For information about enabling and configuring logging, see Log Module.

Field

Description

Download Log

Click this button to download the contents of the log as a file on your computer. In the dialog box, you can open the file or save it. The file can be opened in a text editor such as Notepad.

Clear Log

Click this button to remove all entries from the log.

Display

Choose the type of content to display: All, kernel, or system.

Filter

Enter a keyword to filter the log entries that appear in the viewer. The page will display only the entries that include the keyword.

Diagnostics

The ATA includes two built-in diagnostic tools:

- Ping Test

- Traceroute Test

Ping Test

Use the Administration > Diagnostics > Ping Test page to test connectivity between the ATA and a destination.

STEP 1 Enter the IP address or domain name that you want to ping.

STEP 2 Enter a packet size in bytes. The range is 32 to 65500 bytes.

STEP 3 Choose the number of times to send the ping request (5, 10, or Unlimited).

STEP 4 Click Start to Ping to start the test. After the test is complete, the test results appear on the page. While the ping test is running, you can click Stop to abandon the test.

The test results indicate the number of packets sent and received, the percentage of packet loss, and the round-trip speed.

STEP 5 Click Close to close the test results and display the Ping Test form.

Traceroute Test

Use the Administration > Diagnostics > Traceroute page to view the route between the ATA and a destination.

STEP 1 Enter the IP address or domain name of the destination.

STEP 2 Click Start to Traceroute to start the test. The results appear on the page and are refreshed every 5 seconds. During the test, you can click Stop to abandon the test.

The results display up to 30 hops.

STEP 3 Click Close to close the results and display the Traceroute Test form.

Factory Defaults

Use the Administration > Factory Defaults page to reset the ATA to the default configuration. Alternatively, press and hold the RESET button for 20 seconds. All user-changeable non-default settings will be lost. This may include network and service provider data.

You can perform the following tasks:

- Restore Router Factory Defaults: Choose Yes to remove any custom data (router) settings that you have configured. The default settings will be restored when you click Submit .

- Restore Voice Factory Defaults: Choose Yes . to remove any custom settings that you configured on the Voice pages of the configuration utility. The default settings will be restored when you click Submit .

Firmware Upgrade

Use the Administration > Firmware Upgrade page to upgrade the firmware on the ATA. It is not necessary to upgrade unless you are experiencing problems with the ATA or if the new firmware has a feature that you want to use. Before upgrading the firmware, download the firmware upgrade file for the ATA at: www.cisco.com/go/smallbizvoicegateways

STEP 1 Click Browse and select the location of the upgrade file that you downloaded.

STEP 2 Click the Upgrade button to upgrade the firmware.

CAUTION Upgrading the firmware may take several minutes. Until the process is complete, DO NOT turn off the power, press the hardware reset button, or click the Back button in your current browser.

Configuration Management

Use the Administration > Config Management pages to backup and restore the configuration settings for the ATA.

- Backup Configuration

- Restore Configuration

Backup Configuration

Use the Administration > Config Management > Backup Configuration page to back up the ATA configuration settings to a file. You can then later restore these same settings to the ATA.

Click the Backup button to save the configuration information of the ATA. When the dialog box appears, choose a location where you want to save the .cfg file. Tip: Rename the file with a name that includes the date and time when you did the backup.

Restore Configuration

User the Administration > Config Management > Restore Configuration page to restore the ATA configuration settings from a previous backup. It is recommended that you back up your current configuration settings before you restore a configuration.

STEP 1 Click Browse to locate the .cfg file on your computer.

STEP 2 Click Restore to restore the settings from the selected file.

Reboot

Use the Administration > Reboot page to power cycle the ATA (if necessary) from the configuration utility. Alternatively, accomplish this task by pressing the RESET button.

Click the Reboot button to power cycle the ATA. When the warning message appears, read the information, and then click OK to reboot the ATA, or click Cancel to abandon the operation. The ATA and any connected devices will lose network connectivity during this operation.

Viewing the Status and Statistics

This chapter describes how to view the status and statistics for the ATA. It includes the following sections:

- System Information

- Interface Information

- Internet Status

- Port Statistics (SPA122 only)

- DHCP Server Information (SPA122 only)

System Information

Use the Status > System Information page to view information about the ATA and its current settings.

Field

Description

Model

The model number and product description.

Hardware Version

The hardware version number.

Boot Version

The boot firmware version number.

Firmware Version

The current firmware version.

Recovery Firmware

The version number of the recovery firmware.

WAN MAC Address

The MAC address of the WAN interface.

Host Name

The host name of the ATA.

Domain Name

The domain name of the ATA.

Serial Number

The serial number of the ATA.

Current Time

Time that is set on the ATA.

Interface Information

Use the Status > Interface Information page to view information for the WAN interface (INTERNET port) and on SPA122 only, the LAN interface (ETHERNET port).

Interface List

Field

Description

Interface

The name of the interface: WAN or LAN (SPA122 only).

Connect Type

The type of connection configured for the interface.

IP Address

The IP address of the interface.

Subnet Mask

The subnet mask of the interface.

MAC Address

The MAC address of the interface.

Port List (SPA122 only)

Field

Description

Interface

The name of the interface: WAN or LAN.

TX (pkts)

The number of packets transmitted from this port.

RX (pkts)

The number of packets received by this port.

Status

The status of the port, showing whether the port is connected to a device or disconnected.

Clear TX & RX

Click this button to reset to 0 the count of TX and RX packets.

Internet Status

Field

Description

Link Status

The status of the INTERNET (WAN) interface, showing whether the port is connected or disconnected.

IP Address

The IP address of the INTERNET (WAN) interface.

Netmask

The subnet mask for the INTERNET (WAN) interface.

Gateway

The IP address of the default gateway.

Host Name

The host name of the ATA.

Domain Name

The domain name of the ATA.

MTU Type

The method for setting the MTU: Auto or Manual.

MTU Size

The largest protocol data unit (in bytes) permitted for network transmission.

DNS 1-3 (if applicable)

IP addresses for up to three DNS servers that are used for name resolution.

Port Statistics (SPA122 only)

Field

Description

Input (pkts)

The number of packets received by the port.

Output (pkts)

The number of packets transmitted by the port.

Input Errors

The number of receive errors for incoming traffic.

Input Broadcasts

The number of broadcast messages received by the interface.

Output Broadcasts

The number of broadcast messages sent by the interface.

Input Multicasts

The number of multicast messages received by the interface.

Output Multicasts

The number of multicast messages sent by the interface.

DHCP Server Information (SPA122 only)

Use the Status > DHCP Server Information page to view information about the DHCP server and clients.

DHCP Pool Information

Field

Description

Client Name

The host name of the DHCP client.

IP Address

The IP address leased to the client.

MAC Address

The MAC address of the DHCP client.

Expires Time

The remaining time in the current DHCP lease, shown in HH:MM:SS (hours:minutes:seconds) format. The page is periodically updated with the new value as the timer counts down.

Interface

The interface through which the client is connected.

DHCP Server Details

Field

Description

DHCP Server

The status of the DHCP server: Enabled or Disabled.

IP Address / Mask

The IP address and subnet mask for the ETHERNET (LAN) interface.

DNS Proxy

The setting for the DNS proxy service: Enabled or Disabled.

Maximum DHCP Users

The maximum number of clients that can lease an IP address from the DHCP server.

IP Address Range

The range of IP addresses that can be dynamically assigned by the DHCP server.

Client Lease Time

The maximum amount of time, in minutes, that a client can lease a dynamically assigned IP address.

Static DNS

The IP addresses of up to three DNS servers to be used by DHCP clients.

Option 66

The setting for Option 66, which provides provisioning server address information to hosts requesting this option. The ATA may be set to None (internal), Remote TFTP Server, or Manual TFTP Server.

TFTP Server

The IP address, hostname, or URL of the TFTP server used for provisioning.

Option 67

The configuration/bootstrap filename that is provided to hosts that request this option.

Option 159

The configuration URL that is provided to clients that request this option.

Option 160

The configuration URL that is provided to clients that request this option.

Frequently Asked Questions

Q. I cannot connect to the Internet through the ATA.

STEP 1 Make sure that the ATA is powered on. The Power/Sys LED should be solid green and not flashing.

If the Power LED is flashing, then power off all of your network devices, including the modem, the ATA, and the connected devices. Wait for 30 seconds. Then power on each device in the following order:

- Cable or DSL modem

- ATA

- Connected Devices

STEP 2 Check the cable connections. Ensure that the cable in the INTERNET (WAN) port is securely connected to the device that provides your Internet access, such as your modem or ADSL line. On the Cisco SPA122, check the cable connection for the ETHERNET (LAN) port.

STEP 3 Check the settings on the Network Setup > Internet Settings page. Verify that you entered the settings specified by your Internet Service provider.

Q. I upgraded my firmware and now the ATA is not working properly.

If the ATA is not working properly after an upgrade, you may need to perform a factory reset. Use the Administration > Factory Defaults page to reset the ATA to the default configuration. Alternatively, press and hold the RESET button for 20 seconds. All user-changeable non-default settings will be lost. This may include network and service provider data.

Q. I cannot use the DSL service to connect manually to the Internet.

After you have installed the ATA, it will automatically connect to your service provider’s network, so you no longer need to connect manually.

Q. There is no dial tone, and the Phone 1 or 2 LED is not solid green.

STEP 1 Make sure the telephone is connected to the appropriate port, PHONE 1 or 2.

STEP 2 Disconnect the RJ-11 telephone cable from the PHONE port, and then reconnect it.

STEP 3 Make sure your telephone is set to its tone setting (not pulse).

STEP 4 Make sure your network has an active Internet connection. Try to access the Internet, and check to see if the ATA WAN LED is flashing green. If you do not have a connection, then power off all of your network devices, including the modem, the ATA, and the computers. Wait 30 seconds. Then power on each device in the following order:

- Cable or DSL modem

- ATA

- Computers and other devices

STEP 5 Verify the settings on the Quick Setup page. Verify that you entered the account information and settings required by your service provider. On the Voice > Info page, Line 1 or Line 2 Status section, verify that the Registration State is registered. If the line is not registered, check with your ITSP to determine if additional settings are required.

Q. When I place an Internet phone call, the audio breaks up.

Consider the following possible causes and solutions:

- Network activity

There may be heavy network activity, particularly if you are running a server or using a file sharing program. Try to limit network or Internet activity during Internet phone calls. For example, if you are running a file sharing program, files may be uploaded in the background even though you are not downloading any files, so make sure you exit the program before making Internet phone calls.

- Bandwidth

There may insufficient bandwidth available for your Internet phone call. You may want to test your bandwidth by using one of the bandwidth tests available online. If necessary, access your Internet phone service account and reduce the bandwidth requirements for your service. For more information, refer to the website of your ITSP.

Q. When I open a web browser, I am prompted for a username and password. How can I bypass this prompt?

Launch the web browser and perform the following steps (these steps are specific to Internet Explorer but are similar for other browsers):

STEP 1 Select Tools > Internet Options.

STEP 2 Click the Connections tab.

STEP 3 Select Never dial a connection.

STEP 4 Click OK.

Q. The DSL telephone line does not fit into the ATA WAN (Internet) port.

The ATA does not replace your modem. You need your DSL modem in order to use the ATA. Connect your telephone line to the DSL modem.

Q. My modem does not have an Ethernet port.

If your modem does not have an Ethernet port, then it is a modem for traditional dial-up service. To use the ATA, you need a cable/DSL modem and a high-speed Internet connection.

Q. The ATA does not have a coaxial port for the cable connection.

The ATA does not replace your modem. You need your cable modem in order to use the ATA. Connect your cable connection to the cable modem.

Q. A firmware upgrade (or downgrade) failed.

You can run a firmware recovery procedure to either retry the failed upgrade or install an earlier version of the firmware.

STEP 1 Reboot the ATA.

STEP 2 Connect your computer to the ATA or to a switch that is connected to the ATA.

STEP 3 Start a web browser, and enter the LAN IP address of the ATA. The default LAN IP address is 192.168.15.1.

STEP 4 Log in to the ATA. The default user name and password are both cisco. The password is case sensitive. After you log in, the Recovery Firmware page appears.

STEP 5 Click Browse, select the firmware to install, and then click Start to Upgrade.

Using the IVR for Administration

An IVR system is available to help you to configure and manage your ATA. Use a telephone keypad to select options and to make your entries.

To access the IVR menu:

STEP 1 Connect an analog phone to a PHONE port of the ATA.

STEP 2 Press the star key four times: ****

STEP 3 When challenged for a password, log in as an administrator by using the default administrator's password of 1234# or log in as the PHONE port's user by pressing #.

STEP 4 Enter the code for the desired action. See the IVR Actions table for details.

TIPS:

- Enter the numbers slowly, listening for the audio confirmation before entering the next number.

- After you select an option, press the # (pound) key.

- To exit the menu, hang up the telephone or enter 3948# to exit.

- After entering a value, such as an IP address, press the # (pound) key to indicate that you have finished your selection. Then proceed as needed:

- To save a setting, press 1.

- To review a setting, press 2.

- To re-enter a setting, press 3.

- To cancel your entry and return to the main menu, press * (star).

- While entering a value, you can cancel the changes by pressing the * (star) key twice within half a second. Be sure to press the key quickly, or the * will be treated as a decimal point entry.

- If the menu is inactive for more than one minute, the IVR times out. You will need to re-enter the IVR menu by pressing the star key four times: **** . Your settings take effect after you hang up the telephone or exit the IVR. The ATA may reboot at this time.

- To enter the decimal points in an IP address, press the * (star) key.

For example, to enter the IP address 191.168.1.105, perform the following tasks:

– Press these keys: 191*168*1*105

– Press the # (pound) key to indicate that you have finished entering the IP address.

– Press 1 to save the IP address or press the * (star) key to cancel your entry and return to the main menu.

IVR Actions

IVR Action

Menu Option

Choices and Instructions

Enter IVR Menu

****

Set Internet Addressing Method

101

DHCP: 0

Static IP: 1

PPPoE: Press 2

Check Internet IP Address (INTERNET port)

Set Static IP Address (INTERNET port)

111

Enter the IP address by using numbers on the telephone key pad. Use the * (star) key when entering a decimal point.

Note: This option is available only after you choose Static IP as the Internet Connection Type, through option 101.

Check Network Mask

120

121

To enter the value, press numbers on the telephone key pad. Press the * (star) key to enter a decimal point.

Note: This option is available only after you choose Static IP as the Internet Connection Type, through option 101.

Check Gateway IP Address

Set Gateway IP Address

131

To enter the value, press numbers on the telephone key pad. Press the * (star) key to enter a decimal point.

Note: This option is available only after you choose Static IP as the Internet Connection Type, through option 101.

Check MAC Address

140

Check Primary DNS Server Setting

160

161

To enter the value, press numbers on the telephone key pad. Press the * (star) key to enter a decimal point.

Note: This option is available only after you choose Static IP as the Internet Connection Type, through option 101.

Check INTERNET web server port

SPA122 only: Check LAN IP address (ETHERNET port)

210

Set Line 1 SIP Transport

1911

0: UDP

1: TCP

2: TLS

Check Line 2 SIP Transport

Set Line 2 SIP Transport

1921

0: UDP

1: TCP

2: TLS

Exit IVR

Allow or prevent WAN access to the administration web server

7932

1: Enable

0: Disable

Factory Reset of Unit

WARNING: All non-default settings will be lost. This includes network and service provider data.

73738

(Spells RESET on the phone keypad)

When prompted, press 1 to confirm, or press * (star) to cancel. After you hear “Option successful,” hang up the phone. The ATA reboots.

NOTE: This action is equivalent to Pressing and holding the RESET button for 10 seconds.

Reboot of Voice System

732668

(Spells REBOOT on the phone keypad)

After you hear “Option successful,” hang up the phone. The ATA reboots.

NOTE: This action is equivalent to Pressing and immediately releasing the RESET button.

User Factory Reset of Unit

WARNING: All user-changeable non-default settings will be lost. This may include network and service provider data.

877778

Advanced Options for Voice Services

This appendix provides additional information about configuring advanced options for voice services.

STEP 1

- Optimizing Fax Completion Rates

- Configuring Dial Plans

Optimizing Fax Completion Rates

Issues can occur with fax transmissions over IP networks, even with the T.38 standard, which is supported by the ATA. You can adjust several settings on your ATA to optimize your fax completion rates.

STEP 1 Ensure that you have enough bandwidth for the uplink and the downlink.

- For G.711 fallback, it is recommend to have approximately 100 kbps.

- For T.38, allocate at least 50 kbps.

STEP 2 Click Voice in the menu bar, and then click Line 1 or Line 2 in the navigation tree.

STEP 3 In the Network Settings section, enter the following settings:

- Network Jitter Level : very high

- Jitter Buffer Adjustment : no

STEP 4 In the Supplementary Service Subscription section, enter the following settings:

- Call Waiting Serv : no

- Three Way Call Serv: no

STEP 5 In the Audio Configuration section, enter the following settings to support T.38 fax:

- Preferred Codec : G.711u (USA) or G.711a (rest of the world)

- Use pref. codec only : yes

- Silence Supp Enable : no

- Echo Canc Enable : no

- FAX Passthru Method: ReINVITE

STEP 6 Click Submit to save your settings or click Cancel to abandon the unsaved settings.

STEP 7 If you are using a Cisco media gateway for PSTN termination, disable T.38 (fax relay) and enable fax using modem passthrough. For example:

modem passthrough nse payload-type 110 codec g711ulaw

fax rate disable

fax protocol pass-through g711ulaw

Note: If a T.38 call cannot be set-up, then the call automatically reverts to G.711 fallback.

STEP 8 If you are using a Cisco media gateway, make sure the Cisco gateway is correctly configured for T.38 with the dial peer. For example:

fax protocol T38

fax rate voice

fax-relay ecm disable

fax nsf 000000

no vad

Fax Troubleshooting

If you have problems sending or receiving faxes, complete the following steps:

STEP 1 Verify that your fax machine is set to a speed between 7200 and 14400.

STEP 2 Send a test fax in a controlled environment between two ATAs.

STEP 3 Determine the success rate.

STEP 4 Monitor the network and record the statistics for jitter, loss, and delay.

STEP 5 If faxes fail consistently, capture a copy of the configuration as described below. You can then send this file to Technical Support.

a. In your web browser, enter the path for the configuration file:

http://<ATA_Local_IP_Address>/admin/config.xml&xuser=

<admin_user>&xpassword=<admin_password>

b. On the File menu, choose Save As, and save the file with a filename such as MyConfiguration.xml.

STEP 6 To enable logging, go to the Voice > System page, and set the IP address of your syslog and/or debug server. Set the Debug Level to 3. For more information, see System.

NOTE: You can also capture data using a sniffer trace.

STEP 7 Identify the type of fax machine connected to the ATA.

STEP 8 Contact technical support:

- If you are an end user of VoIP products, contact the reseller or service provider that supplied the equipment.

- If you are an authorized Cisco partner, contact Cisco technical support. For contact options, see: www.cisco.com/go/sbsc

Configuring Dial Plans

Dial plans determine how dialed digits are interpreted and transmitted. They also determine whether the dialed number is accepted or rejected. You can use a dial plan to facilitate dialing or to block certain types of calls such as long distance or international. This section includes information that you need to understand dial plans, as well as procedures for configuring your own dial plans.

To edit a dial plan, click Voice on the menu bar, and then click Line 1 or Line 2 in the navigation tree. Scroll down to the Dial Plan section, and then enter the digit sequences in the Dial Plan field. For more information and examples, see the following topics:

- Digit Sequences

- Acceptance and Transmission of the Dialed Digits

- Dial Plan Timer (Off-Hook Timer)

- Interdigit Long Timer (Incomplete Entry Timer)

- Interdigit Short Timer (Complete Entry Timer)

- Resetting the Control Timers

Digit Sequences

A dial plan contains a series of digit sequences, separated by the pipe character: |

The entire collection of sequences is enclosed within parentheses. Each digit sequence within the dial plan includes 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.

Digit Sequence

Function

0 1 2 3 4 5 6 7 8 9 0

* #

Enter any of these characters to represent a key that the user must press on the phone keypad.

x

Enter x to represent any character on the phone keypad.

[sequence]

Enter characters within square brackets to create a list of accepted key presses. The user can press any one of the keys in the list.

- Numeric range: For example, you would enter [2-9] to allow the user to press any one digit from 2 through 9.

- Numeric range with other characters: For example, you would enter [35-8*] to allow the user to press 3, 5, 6, 7, 8, or *.

.

(period)

Enter a period for element repetition. The dial plan accepts zero or more entries of the digit. For example, 01. allows users to enter 0, 01, 011, 0111, and so on.

<dialed:substituted>

Use this format to indicate that certain dialed digits are replaced by other characters when the sequence is transmitted. The dialed digits can be zero or more characters.

EXAMPLE 1: <8:1650>xxxxxxx

When the user presses 8 followed by a seven digit number, the system automatically replaces the dialed 8 with 1650. If the user dials 85550112, the system transmits 16505550112.

EXAMPLE 2: <:1>xxxxxxxxxx

In this example, no digits are replaced. When the user enters a 10-digit string of numbers, the number 1 is added at the beginning of the sequence. If the user dials 9725550112, the system transmits 19725550112.

,

(comma)

Enter a comma between digits to play an “outside line” dial tone after a user-entered sequence.

EXAMPLE: 9, 1xxxxxxxxxx

An “outside line” dial tone is sounded after the user presses 9, and the tone continues until the user presses 1.

!

(exclamation point)

Enter an exclamation point to prohibit a dial sequence pattern.

EXAMPLE: 1900xxxxxxx!

The system rejects any 11-digit sequence that begins with 1900.

*xx

Enter an asterisk to allow the user to enter a 2-digit star code.

S0 or L0

Enter S0 to reduce the short inter-digit timer to 0 seconds, or enter L0 to reduce the long inter-digit timer to 0 seconds.

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.

EXAMPLE: ([1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9, 011xxxxxx. | 0 | [49]11 )

NOTE Red text is used to highlight the elements that are explained in the examples.

- Extensions on your system

( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9, 011xxxxxx. | 0 | [49]11 )

[1-8]xx Allows a user dial any three-digit number that starts with the digits 1 through 8. If your system uses four-digit extensions, you would instead enter the following string: [1-8]xxx

- Local dialing with seven-digit number

( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9, 011xxxxxx. | 0 | [49]111)

9, xxxxxxx After a user presses 9, an external dial tone sounds. The user can then dial any seven-digit number, as in a local call.

- Local dialing with 3-digit area code and a 7-digit local number

( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8,<:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx !| 9, 011xxxxxx. | 0 | [49]11 )

9, <:1>[2-9]xxxxxxxxx This example is useful where a local area code is required. After a user presses 9, an external dial tone sounds. The user must enter a 10-digit number that begins with a digit 2 through 9. The system automatically inserts the 1 prefix before transmitting the number to the carrier.

- Local dialing with an automatically inserted 3-digit area code

( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9, 011xxxxxx. | 0 | [49]11 )

8, <:1212>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 system automatically inserts the 1 prefix and the 212 area code before transmitting the number to the carrier.

- U.S. long distance dialing

( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx |8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9,011xxxxxx. | 0 | [49]11 )

9, 1 [2-9] xxxxxxxxx After the user presses 9, an external dial tone sounds. The user can enter any 11-digit number that starts with 1 and is followed by a digit 2 through 9.

- Blocked number

( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx |8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! |9, 011xxxxxx. | 0 | [49]11 )

9, 1 900 xxxxxxx ! This digit sequence is useful if you want to prevent users from dialing numbers that are associated with high tolls or inappropriate content, such as 1-900 numbers in the United States. After the user press 9, an external dial tone sounds. If the user enters an 11-digit number that starts with the digits 1900, the call is rejected.

- U.S. international dialing

( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9, 011xxxxxx. | 0 | [49]11 )

9, 011xxxxxx. After the user presses 9, an external dial tone sounds. The user can enter any number that starts with 011, as in an international call from the United States.

- Informational numbers

( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9, 011xxxxxx. | 0 | [49]11 )

0 | [49]11 This example includes two digit sequences, separated by the pipe character. The first sequence allows a user to dial 0 for an operator. The second sequence allows the user to enter 411 for local information or 911 for emergency services.

Acceptance and Transmission of the Dialed Digits

When a user dials a series of digits, each sequence in the dial plan is tested as a possible match. The matching sequences form a set of candidate digit sequences. As more digits are entered by the user, the set of candidates diminishes until only one or none are valid. When a terminating event occurs, the ATA either accepts the user-dialed sequence and initiates a call, or else rejects the sequence as invalid. The user hears the reorder (fast busy) tone if the dialed sequence is invalid.

The following table explains how terminating events are processed.

Terminating Event

Processing

The dialed digits do not match any sequence in the dial plan.

The number is rejected.

The dialed digits exactly match one sequence in the dial plan.

- If the sequence is allowed by the dial plan, the number is accepted and is transmitted according to the dial plan.

- If the sequence is blocked by the dial plan, the number is rejected.

A timeout occurs.

The number is rejected if the dialed digits are not matched to a digit sequence in the dial plan within the time specified by the applicable interdigit timer.

- The Interdigit Long Timer applies when the dialed digits do not match any digit sequence in the dial plan. Default setting: 10 seconds

- The Interdigit Short Timer applies when the dialed digits match one or more candidate sequences in the dial plan. Default setting: 3 seconds

The user presses the # key.

- If the sequence is complete and is allowed by the dial plan, the number is accepted and is transmitted according to the dial plan.

- If the sequence is incomplete or is blocked by the dial plan, the number is rejected.

Dial Plan Timer (Off-Hook Timer)

You can think of the Dial Plan Timer as “the off-hook timer.” This timer starts counting when the phone goes off hook. If no digits are dialed within the specified number of seconds, the timer expires and the null entry is evaluated. Unless you have a special dial plan string to allow a null entry, the call is rejected. Default setting: 5

Syntax for the Dial Plan Timer

(Ps<:n> | dial plan )

- s: The number of seconds; if no number is entered after P, the default timer of 5 seconds applies.

- n: (optional): The number to transmit automatically when the timer expires; you can enter a valid number. No wildcard characters are allowed because the number will be transmitted as shown. If you omit the number substitution, <:n>, then the user hears a reorder (fast busy) tone after the specified number of seconds.

Examples for the Dial Plan Timer

NOTE Red text is used to highlight the elements that are explained in the examples.

- Allow more time for users to start dialing after taking a phone off hook.

( P9 | (9,8<:1408>[2-9]xxxxxx | 9,8,1[2 9]xxxxxxxxx | 9,8,011xx. | 9,8,xx.|[1-8]xx)

P9 After 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.

xx This code allows the entry of one or more digits. Do not use a single x, allowing 0 or more digits. This setting will produce unwanted results especially if you are deploying timers.

- Create a hotline for all sequences on the System Dial Plan

( P9 <:23> | (9,8<:1408>[2-9]xxxxxx | 9,8,1[2-9]xxxxxxxxx | 9,8,011xx. | 9,8,xx.|[1-8]xx)

P9 <:23> After taking the phone off hook, a user has 9 seconds to begin dialing. If no digits are pressed within 9 seconds, the call is transmitted automatically to extension 23.

- Create a hotline on a line button for an extension

(P0 <:1000>)

With the timer set to 0 seconds, the call is transmitted automatically to the specified extension when the phone goes off hook.

Interdigit Long Timer (Incomplete Entry Timer)

You can think of this timer as the “incomplete entry” timer. This timer measures the interval between dialed digits. It applies as long as the dialed digits do not match any digit sequences in the dial plan. Unless the user enters another digit within the specified number of seconds, the entry is evaluated as incomplete, and the call is rejected. Default setting: 10 seconds

NOTE This section explains how to edit a timer as part of a dial plan. Alternatively, you can modify the Control Timer that controls the default interdigit timers for all calls. See Resetting the Control Timers.

Syntax for the Interdigit Long Timer

L:s, ( dial plan )

s: The number of seconds; if no number is entered after L:, the default timer of 5 seconds applies. The timer sequence appears to the left of the initial parenthesis for the dial plan.

Example for the Interdigit Long Timer

L:15, (9,8<:1408>[2-9]xxxxxx | 9,8,1[2-9]xxxxxxxxx | 9,8,011xx. | 9,8,xx.|[1-8]xx)

L:15, This dial plan allows the user to pause for up to 15 seconds between digits before the Interdigit Long Timer expires.

Interdigit Short Timer (Complete Entry Timer)

You can think of this timer as the “complete entry” timer. This timer measures the interval between dialed digits. It applies when the dialed digits match at least 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. Default setting: 3 seconds

Syntax for the Interdigit Short Timer

SYNTAX 1: S:s, ( dial plan )

Use this syntax to apply the new setting to the entire dial plan within the parentheses.

SYNTAX 2: sequence Ss

Use this syntax to apply the new setting to a particular dialing sequence.

s: The number of seconds; if no number is entered after S, the default timer of 5 seconds applies.

Examples for the Interdigit Short Timer

Set the timer for the entire dial plan.

S:6,(9,8<:1408>[2-9]xxxxxx | 9,8,1[2-9]xxxxxxxxx | 9,8,011xx. | 9,8,xx.|[1-8]xx)

S:6, While entering a number with the phone off hook, a user can pause for up to 15 seconds between digits before the Interdigit Short Timer expires.

Set an instant timer for a particular sequence within the dial plan.

(9,8<:1408>[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.

Resetting the Control Timers

You can use the following procedure to reset the default timer settings for all calls.

NOTE To edit a timer setting only for a particular digit sequence or type of call, you can edit the dial plan. See Digit Sequences.

STEP 1 Log in to the configuration utility. If prompted, enter the administrative logon provided by the Service Provider. The default username and password are both admin.

STEP 2 Under the Voice menu, click Regional.

STEP 3 In the Control Timer Values section, enter the desired values in the Interdigit Long Timer field and the Interdigit Short Timer field. See the definitions at the beginning of this section.

Where to Go From Here

Cisco provides a wide range of resources to help you and your customer obtain the full benefits of the ATA.

Support

Cisco Small Business Support Community

www.cisco.com/go/smallbizsupport

Online Technical Support and Documentation (Login Required)

www.cisco.com/support

Cisco Small Business Support and Resources

www.cisco.com/go/smallbizhelp

Downloads and Documentation

Firmware

www.cisco.com/go/software

Cisco Small Business Voice Gateways Documentation

www.cisco.com/go/smallbizvoicegateways

Open Source Documentation

Follow the Release Notes link at www.cisco.com/go/smallbizvoicegateways

Cisco Small Business

Cisco Partner Central for Small Business (Partner Login Required)

www.cisco.com/web/partners/sell/smb

Cisco Small Business Home

www.cisco.com/smb

Getting Started

Thank you for choosing the Cisco SPA100 Series Phone Adapters. This chapter provides more information about the features of the product and the web-based configuration utility.

Feature Overview

Product Features

Connecting the Equipment

Configuration and Management of the ATA

Mounting the ATA

Elements of the User Interface

Feature Overview

Cisco SPA100 Series Analog Telephone Adapters (ATAs) provide your standard analog phones with access to Internet phone services through two standard telephone RJ-11 phone ports. The ATA connects to the Internet through a broadband (DSL or cable) modem or router. The ATA can be used with an on-site call-control system or an Internet-based call-control system.

The 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. An ATA maintains the state of each call it terminates and reacts appropriately to user input events (such as on/off hook or hook flash). The ATAs use the Session Initiation Protocol (SIP) open standard so there is little or no involvement by a “middle-man” server or media gateway controller. SIP allows inter-operation with all ITSPs that support SIP.

The system supports four simultaneous calls, including “active” calls and “on-hold” calls. A phone can handle one on-hold call and one active call simultaneously.

Understanding Voice Service Operations

The ATA allows calls to be made by using SIP-based Voice-over-IP (VoIP) services and traditional telephone Public Switched Telephone Network (PSTN) services. Calls can be placed and received by using an analog phone or fax machine.

The ATA maintains the state of each call and makes the proper reaction to user input events (such as on/off hook or hook flash). Because the ATA uses the Session Initiation Protocol (SIP), it is compatible with most Internet Telephony Service Provider (ITSP) offerings.

ATA Voice Features

The ATA can be custom provisioned within a wide range of configuration parameters. The following sections describe the factors that contribute to voice quality:

- Supported Codecs

- SIP Proxy Redundancy

- Other ATA Voice Features

Supported Codecs

Codec

Description

G.711 (A-law and mu-law)

Very low complexity codecs that support uncompressed 64 kbps digitized voice transmissions at one through ten 5 ms voice frames per packet. These codecs provide the highest narrow-band voice quality and uses the most bandwidth of any of the available codecs.

G.726-32

Low complexity codec that supports compressed 32 kbps digitized voice transmission at one through ten 10 ms voice frames per packet. This codec provides high voice quality.

G.729a

ITU G.729 voice coding algorithm used to compress digitized speech. G.729a is a reduced complexity version of G.729 requiring about half the processing power of G.729. The G.729 and G.729a bit streams are compatible and interoperable, but not identical.

SIP Proxy Redundancy

In typical commercial IP Telephony deployments, all calls are established through a SIP proxy server. A typical SIP proxy server can handle thousands of subscribers. It is important that a backup server be available so that an active server can be temporarily switched out for maintenance. The ATA supports the use of backup SIP proxy servers (through DNS SRV) so that service disruption is minimized.

An easy way to support proxy redundancy is to configure your DNS server with a list of SIP proxy addresses. The ATA can be instructed to contact a SIP proxy server in a domain named in the SIP message. The ATA consults the DNS server to get a list of hosts in the given domain that provide SIP services. If an entry exists, the DNS server returns an SRV record that contains a list of SIP proxy servers for the domain, with their host names, priority, listening ports, and so on. The ATA tries to contact the list of hosts in the order of their stated priority.

If the ATA is currently using a lower priority proxy server, it periodically probes the higher priority proxy to see whether it is online, and switches back to the higher priority proxy when possible. You can use the default settings or configure the Proxy Redundancy Method in the Proxy and Registration section of the Line 1 and Line 2 Settings (PHONE 1 and PHONE 2) page

Other ATA Voice Features

- 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 average bandwidth required for a single call. VAD distinguishes between speech and non-speech signals, and Silence Suppression removes the natural silences that occur in a conversation. Therefore the IP bandwidth is used only to transmit speech. Comfort Noise Generation provides artificially-generated background white noise (sounds) to reassure callers that their calls are still connected during the silent periods. You can enable this feature in the Audio Configuration section of the Line 1 and Line 2 Settings (PHONE 1 and PHONE 2) page.

- Modem and Fax Pass-Through

- Modem pass-through mode can be triggered by predialing the Vertical Service Activation Code for the Modem Line Toggle Code. You can configure this setting in the Vertical Service Activation Codes section of the Regional page.

- FAX pass-through mode is triggered by the detection of a CED/CNG tone or an NSE event.

- Echo canceller is automatically disabled for Modem passthrough mode.

- Echo canceller is disabled for FAX pass-through if the parameter FAX Disable ECAN (Line 1 or 2 tab) is set to “yes” for that line (in that case FAX pass-through is the same as Modem pass-through)

- Call waiting and silence suppression are automatically disabled for both FAX and Modem pass-through. In addition, out-of-band DTMF transmission is disabled during modem or fax passthrough.

- Adaptive Jitter Buffer

The ATA can buffer incoming voice packets to minimize the impact of variable network delays. This process is known as jitter buffering. The size of the jitter buffer adjusts to changing network conditions. The ATA has a Network Jitter Level control setting for each line of service. The jitter level determines how aggressively the ATA tries to shrink the jitter buffer over time to achieve a lower overall delay. If the jitter level is higher, it shrinks more gradually. If jitter level is lower, it shrinks more quickly. You can use the default settings or configure this feature in the Network Settings section of the Line 1 and Line 2 Settings (PHONE 1 and PHONE 2) page.

- Adjustable Audio Frames Per Packet

This feature allows the user to set the number of audio frames contained in one RTP packet. Packets can be adjusted to contain from 1–10 audio frames. Increasing the number of packets decreases the bandwidth utilized, but it also increases delay and may affect voice quality. You can configure this setting in the RTP Parameters section of the SIP page.

- DTMF Relay

The ATA may relay DTMF digits as out-of-band events to preserve the fidelity of the digits. This can enhance the reliability of DTMF transmission required by many IVR applications such as dial-up banking and airline information. You can configure this setting in the RTP Parameters section of the SIP page.

- Call Progress Tones

The ATA has configurable call progress tones. Call progress tones are generated locally on the ATA so that an end user is advised of status (such as ringback) Parameters for each type of tone (for instance a dial tone played back to an end user) may include frequency and amplitude of each component, and cadence information. You can keep the default settings or configure these tones in the Call Progress Tones section of the Regional page.

- Call Progress Tone Pass Through

This feature allows the user to hear the call progress tones (such as ringing) that are generated from the far-end network.

- Echo Cancellation

Impedance mismatch between the telephone and the IP Telephony gateway phone port can lead to near-end echo. The ATA has a near-end echo canceller that compensates for impedance mismatch. The ATA also implements an echo suppressor with Comfort Noise Generator (CNG) so that any residual echo is not noticeable. This feature is enabled by default. You can configure this setting in the Audio Configuration of the Line 1 and Line 2 Settings (PHONE 1 and PHONE 2) page.

- Hook Flash Events

The ATA can signal hook flash events to the proxy during a connected call. This feature can be used to provide advanced mid-call services with third-party-call control.

- Depending on the features that the service provider offers using third-party-call-control, you may need to disable Call Waiting Service, Three Way Conference Service, or Three Way Call Service to correctly signal a hook flash event to the softswitch. You can configure these settings in the Supplementary Service Subscription section of the Line 1 and Line 2 Settings (PHONE 1 and PHONE 2) page.

- You can configure the length of time allowed for detection of a hook flash by adjusting the Hook Flash Timer parameter in the Control Timer Values section of the SIP page.

- Configurable Dial Plan with Interdigit Timers

The ATA has three configurable interdigit timers: an initial timeout signaling that a phone is taken off hook, a long timeout signaling the end of a dialed string, and a short timeout, signaling that more digits are expected. For more information, see Configuring Dial Plans.

- Polarity Control

The ATA allows the polarity to be set when a call is connected and when a call is disconnected. This feature is required to support some pay phone system and answering machines. You can configure these settings in the FXS Port Polarity Configuration section of the Line 1 and Line 2 Settings (PHONE 1 and PHONE 2) page.

- Calling Party Control

Calling Party Control (CPC) signals to the called party equipment that the calling party has hung up during a connected call by momentarily removing the voltage between the tip and the ring. This feature is useful for auto-answer equipment. You can configure these settings in the Control Timer Values section of the Regional page.

- Event Logging

You can enable logging and select the relative priority of events to be logged. The information can be sent to a Syslog Server. You can configure the syslog and debug settings in the Miscellaneous Settings section of the System page.

- Encryption of SIP messages using SIP over TLS

You can enable SIP over Transport Layer Security (TLS) to encrypt the SIP messages between the service provider and the your business. SIP over TLS relies on the widely-deployed and standardized TLS protocol to encrypt the signaling messages. You can configure the SIP Transport parameter in the SIP Settings section of the Line 1 and Line 2 Settings (PHONE 1 and PHONE 2) page.

- Secure Calling using SRTP