Table of Contents

Product Overview and Deployment Guidelines

Configure Your System for ITSP Interoperability

Configure Voice Services

Advanced Voice Fields

Data Fields

Troubleshooting

Environmental Specifications for the WRP500

Where to Go From Here

Title Page - DO NOT PUBLISH

Product Overview and Deployment Guidelines

This chapter describes the features and benefits of the WRP500, describes deployment scenarios, and offers guidelines to help you plan your network.

- WRP500 Features and Benefits

- Deployment Models

- Local Area Network Guidelines

- Special Requirements for Voice Deployments

- WRP500 Maintenance Operations

- Remote Provisioning

WRP500 Features and Benefits

With a variety of features, the WRP500 offers the benefits of five devices in one:

- Router: The WRP500 is a broadband router with a robust security firewall to protect your network.

- Switch: The WRP500 includes a built-in, 4-port, full-duplex, 10/100/1000M Ethernet switch to connect computers, printers, and other equipment directly or to attach additional hubs and switches. Advanced Quality of Service functionality ensures that you can prioritize traffic for data, voice, and video applications.

- Analog Telephone Adapter: The WRP500 includes a two-port Analog Telephone Adapter (ATA) that allows you to connect your analog phones or fax machines to your configured Internet telephone service. Two traditional phone lines also can be connected for support of legacy phone numbers and fax numbers.

- Wireless Access Point: The WRP500 has an integrated 802.11ac/b/g/n wireless access point that secures your communications with WEP, WPA, and WPA2 security protocols. It is preconfigured to support two wireless networks: one for transferring general data, such as data from a connected PC; and another for transferring data from voice devices, such as audio or fax data.

- Mobile Broadband Router: When you attach a compatible Mobile Broadband Modem to the USB port, the WRP500 allows multiple Wi-Fi and Ethernet devices to share a mobile broadband connection. This feature also can be used to provide continuous Internet service by providing automatic failover to the mobile network when the primary Internet connection is unavailable. For the latest copy of the USB Modem Compatibility List, visit the following URL: http://www.cisco.com/c/en/us/products/unified-communications/wrp500-wireless-g-broadband-router-2-phone-ports/index.html

Note Because this device has many unique functions, the administrative tasks for the WRP500 may be different from corresponding tasks on other Cisco Small Business routers, switches, and ATAs. Administrators should refer to this guide for the proper procedures for installation, configuration, and management of the WRP500.

Deployment Models

The versatility of the WRP500 makes it useful for a variety of deployments:

- WRP500 Deployment in a Basic Network

- WRP500 Deployment with a Wireless Guest Network

- WRP500 Deployment with Mobile Broadband

WRP500 Deployment in a Basic Network

In this scenario, the WRP500 is deployed in a small business that has a basic network configuration.

- The WRP500 is preconfigured by the Service Provider to act as the edge device that routes traffic between the small business network and the Service Provider network.

Note The WRP500 may be configured as an edge device or can be connected to another device that provides access to the Service Provider network.

- The WRP500 connects computers to the Internet. Computers may be connected by network cables or may operate wirelessly. All computers have access to the printer on the local network.

- An analog phone and a fax machine are connected to the WRP500 phone ports and have access to the configured Voice over IP services.

WRP500 Deployment with a Wireless Guest Network

In this example, the WRP500 is deployed in an Internet cafe.

- The WRP500 is connected to a cable modem that provides Internet access.

Note The WRP500 may be configured as an edge device or can be connected to another device that provides access to the Service Provider network.

- In the private network, a computer is connected to the WRP500 by an Ethernet cable. The manager also has a laptop computer that can be used wirelessly from anywhere on the premises through the main wireless network, SSID1. The manager and employees who use SSID1 have access to the printer. If desired, a wireless phone can also connect to this network for business use.

- An analog phone and a fax machine are in the private network. The WRP500 is configured for Internet telephone service.

- The WRP500 is configured with a guest network, SSID2, that enables the business to provide its customers with a free wireless hotspot for their laptop computers and other mobile devices. Because this network is separate from the main wireless network, customers have no access to the manager’s computer, printer, or telephone service.

WRP500 Deployment with Mobile Broadband

When a compatible mobile broadband modem connects to the USB port, the WRP500 can connect to a mobile broadband network. The mobile network can be the primary network or can serve as a backup network to ensure continuous Internet connectivity. Consider the scenarios that follow.

Mobile Office That Uses the Mobile Network for Internet Access

In this example, a team has set up a temporary network at a construction site. Team members have laptop computers and Wi-Fi phones that share a mobile broadband connection for Internet access. All computers can connect to the printer on the local network. If a Virtual Private Network (VPN) tunnel is configured on the laptop computer, team members also can securely connect to resources at the main office (not illustrated).

Basic Office Deployment That Uses the Mobile Network as a Backup Connection

In this example, the business has the same network as illustrated in the “WRP500 Deployment in a Basic Network” section. However, this business has the added benefit of using the mobile broadband network as a backup network to ensure continuous Internet connectivity. In the event that the Internet connection fails, the WRP500 fails over to the configured mobile network. When the Internet connection becomes available, the WRP500 recovers the connection.

Local Area Network Guidelines

This section offers guidelines for setting up your Local Area Network (LAN).

Note As you design your network, be aware that the WRP500 is intended for deployment in a very small business. The router is designed to handle the data, voice, and video traffic that is expected by office personnel who use the Internet to find data, conduct phone conversations, transmit email, and participate in videoconferences. For large-scale operations with heavy data, voice, and video requirements, consider other models of Cisco Small Business routers.

Power, Cabling, and Telephone Lines

- AC outlets: Ensure that an AC outlet is available for every network device that requires AC power.

– The WRP500 requires power, and Ethernet switches (optional) require power.

– Some analog telephones require AC power.

- Ethernet cabling: If an Internet access device is present, you need to connect it to the WRP500 with an Ethernet cable. You also need Ethernet cable for any devices that do not have wireless connectivity. Ethernet cables that are UTP Cat5e or better are recommended.

- UPS: It is strongly recommended that you included an Uninterrupted Power Supply (UPS) mechanism in your network to ensure continuous operation during a power failure. Connect all essential devices, including the Internet access device, the WRP500, and the Ethernet switch (if present).

Basic Services and Equipment

The following basic services and equipment are required:

- An integrated access device or modem for broadband access to the Internet

- Business grade Internet service

- Internet Telephony Service Provider (ITSP) for Voice over IP (VoIP) telephone service that supports a “bring your own device” model

- A computer with Microsoft Windows for system configuration

Special Requirements for Voice Deployments

Voice deployments have special requirements that you must meet to ensure voice quality.

- Bandwidth for Voice Deployments

- NAT Mapping for Voice over IP Deployments

- Local Area Network Design for Voice Deployments

Bandwidth for Voice Deployments

You can choose from several types of broadband access technologies to provide symmetric or asymmetric connectivity to a small business. These technologies vary on the available bandwidth and on the quality of service. For voice deployments, it is generally recommended that you use broadband access with a Service Level Agreement that provides quality of service. If a Service Level Agreement with regard to the broadband connection quality of service is not in place, the downstream audio quality may be affected negatively under heavy load conditions (bandwidth utilization beyond 80%).

To eliminate or minimize this effect, Cisco recommends one of the following actions:

- For broadband connections with a bandwidth lower than 2 Mbps, perform the call capacity calculations by assuming a bandwidth value of 50% of the existing broadband bandwidth. For example, in the case of a 2 Mbps uplink broadband connection, assume 1 Mbps. Limit the uplink bandwidth in the Integrated Access Device to this value. This setting helps to maintain utilization levels below 60%, and thus reduces jitter and packet loss.

- Use an additional broadband connection for voice services only. A separate connection is required when the broadband connection services do not offer quality of service and when it is not possible to apply the above mentioned utilization mechanism.

The available connection bandwidth determines the maximum number of simultaneous calls that the system can support with the appropriate audio quality. Use this information to determine the maximum number of simultaneous VoIP connections that the system can support.

Note Some ITSP SIP trunk services limit the maximum number of simultaneous calls. Please check with your Service Provider to understand the maximum number of simultaneous calls that each SIP trunk supports.

The following table provides the approximate bandwidth budget for different codecs.

Note The Cisco WRP500 supports only the G.711 and G.729 codecs.

Codec

Approximate Bandwidth Budget for Each Side of Conversation

2 Calls

4 Calls

6 Calls

8 Calls

G.711

128 kbps

256 kbps

512 kbps

768 kbps

1024 kbps

G.729

16 kbps

32 kbps

64 kbps

96 kbps

128 kbps

For more information about bandwidth calculation, refer to the following web sites:

www.erlang.com/calculator/lipb/

www.bandcalc.com/

NAT Mapping for Voice over IP Deployments

Network Address Translation (NAT) is the function that allows multiple devices in your small business network to share one external (public) IP address that you receive from your Internet Service Provider. Voice over IP can co-exist with NAT only when some form of NAT traversal is provided.

Some Internet Telephone Service Providers (ITSPs) provide NAT traversal, but some do not. For voice deployments, it is strongly recommended that you choose an ITSP that supports NAT mapping through a Session Border Controller.

If your ITSP does not provide NAT mapping through a Session Border Controller (the preferred method), you have these options for providing NAT traversal on your WRP500:

- Deploy an edge device that has a SIP ALG (Application Layer Gateway). The Cisco Small Business WRV200 is suited for this purpose, but other SIP-ALG routers can be used. If your Internet Service Provider provides the edge device, check with your provider to determine whether the router has a SIP ALG.

- Configure NAT mapping with the EXT IP setting. This option requires that you have (1) a static external (public) IP address from your Internet Service Provider and (2) an edge device with a symmetric NAT mechanism. If the WRP500 is the edge device, the second requirement is met. For more information about the EXT IP setting, see the “NAT Support Parameters section” section.

- Configure Simple Traversal of UDP through NAT (STUN). This option requires that you have (1) a dynamic external (public) IP address from your service provider, (2) a computer that is running STUN server software, and (3) an edge device with an asymmetric NAT mechanism. If the WRP500 is the edge device, the third requirement is not met. For more information about the STUN Enable setting and the STUN Test Enable setting, see the “NAT Support Parameters section” section.

Local Area Network Design for Voice Deployments

Use these guidelines to manage the LAN setup for voice deployments:

- Ensure that all telephones are located in the same local area network subnet.

- Configure your WRP500 as a DHCP server for the purpose of easily adding network devices to the system. Ensure that the DHCP server can assign enough IP addresses to serve the devices that you need to connect to your network.

- Use stable DNS server addresses for URL name resolution. Your Internet Service Provider can provide the primary and secondary DNS server IP addresses.

- If you need to connect more than four network devices directly (other than wireless devices), you need to connect an Ethernet switch to the WRP500. For voice deployments, Cisco recommends use of the SLMxxxP, SRWxxxP and SRWxxxMP switch product families. The SLM224P is a popular choice. For more information about these switches, visit the following URL: www.cisco.com/cisco/web/solutions/small_business/products/routers_switches/index.html

- If you use an Ethernet switch, configure it to ensure voice quality. These settings are recommended:

– Enable Port Fast and Spanning Tree Protocol on the ports to which your voice devices are connected. Cisco phones are capable of rebooting in a few seconds and will attempt to locate network services while a switch port is being blocked by STP after it senses a device reboot. If you enable Port Fast, the network will be available to the phones when it is needed. If the switch does not provide a way to enable Port Fast, you must disable Spanning Tree Protocol.

– In the administrative web pages for the switch, enable QoS and choose DSCP as the Trust Mode.

WRP500 Maintenance Operations

Due to its unique functions, the WRP500 has unique maintenance operations as compared to other Cisco Small Business IP telephony devices.

- Remote Management: For security purposes, remote management is disabled by default.

– When you first configure the WRP500, connect your administrative computer directly to one of the LAN ports and enter the default static IP address into your web browser to log on to the configuration utility.

Note The default LAN IP address of the WRP500 is 192.168.15.1. If another device on the network has the same IP address, the WRP500 takes the address 192.168.16.1. To modify the Local IP Address, go to the Interface Setup tab > LAN > DHCP Server section.

If you are using the IVR, be aware that this address is NOT the address that the 110 option of the IVR reports. The device does not respond to the 110 option address.

– To enable web access and wireless access to the configuration utility, use the Administration tab > Web Access Management section.

- DHCP Server: The DCHP server on LAN ports is enabled by default. This setting is on the Interface Setup tab > LAN > DHCP Server section.

- System Logging: To enable system logging, be aware that two sets of system logs exist: one for the data (router) functions and another for the voice functions.

– Data (router) logging: See the Administration tab > Log page.

– Voice logging: See the Voice tab > System page, Miscellaneous Settings section.

- Factory Reset: To reset your WRP500 to the factory default settings, reset the data (router) settings and the voice settings separately.

Factory Reset of Data (Router) Settings

Use one of the following methods:

- Option 1: Log on to the configuration utility, then click Administration > Factory Defaults . Next to Restore Router Factory Defaults , click Yes . Then click Submit to begin the operation.

- Option 2: Press and hold the reset button located on the rear panel for approximately ten seconds.

Factory Reset of Voice Settings

Use one of the following methods:

- Option 1: Log on to the configuration utility, then click Administration tab > Factory Defaults. Next to Restore Voice Factory Defaults , click Yes . Then click Submit to begin the operation.

- Option 2: Connect an analog phone to the Phone 1 or Phone 2 port. Press **** to access the Interactive Voice Response menu. After you hear the greeting, press 73738 for factory reset. Listen to the prompts, then press 1 to confirm or * to cancel.

Remote Provisioning

Like other Cisco Small Business IP Telephony Devices, the WRP500 provides for secure provisioning and remote upgrade. Provisioning is achieved through configuration profiles that are transferred to the device via TFTP, HTTP, or HTTPS. To configure Provisioning, go to the Provisioning tab in the Configuration Utility.

For complete details, see the Provisioning Guide at the following URL:

http://www.cisco.com/c/dam/en/us/td/docs/voice_ip_comm/csbpvga/ata/provisioning/guide/Provisioning.pdf

Upgrade URL

Remote firmware upgrade is achieved via TFTP or HTTP/HTTPS. Remote upgrades are initiated by causing the WRP500 to request the upgrade firmware image by providing a URL for the WRP500 to retrieve the firmware.

Note The Upgrade/Resync/Reboot URL works only after the administrator logs in to the web GUI.

Note If the value of the Upgrade Enable parameter in the Provisioning page is No, you cannot upgrade the WRP500 even if the web page indicates otherwise.

The syntax of the Upgrade URL is as follows:

http:// WRP500_ip_address /admin/upgrade?[protocol://][server-name[:port]][/firmware-pathname]

HTTP, HTTPS, and TFTP are supported for the upgrade operation.

If no protocol is specified, TFTP is assumed.

If no port specified, the default port of the protocol is used (69 for TFTP, 80 for HTTP, or 443 for HTTPS).

The f irmware-pathname is typically the file name of the binary that is located in a directory on the TFTP, HTTP, or HTTPS server. If no firmware-pathname is specified, /spa.bin is assumed, as in the following example:

http://192.168.2.217/amin/upgrade?tftp://192.168.2.251/spa.bin

Resync URL

The WRP500 can be configured to automatically resync its internal configuration state to a remote profile periodically and on power up. The automatic resyncs are controlled by configuring the desired profile URL into the device.

Note The Upgrade/Resync/Reboot URL works only after the administrator logs in to the web GUI.

The Resync URL lets you force the WRP500 to do a resync to a profile specified in the URL, which can identify either a TFTP, HTTP, or HTTPS server. The syntax of the Resync URL is as follows:

http:// WRP500_ip_address /admin/resync?[[protocol://][server-name[:port]]/profile-pathname]

Note The WRP500 resyncs only when it is idle.

If no port is specified, the default port is used (69 for TFTP, 80 for HTTP, and 443 for HTTPS).

The profile-path is the path to the new profile with which to resync, for example:

http://192.168.2.217/admin/resync?tftp://192.168.2.251/spaconf.xml

Reboot URL

The Reboot URL lets you reboot the WRP500. The Reboot URL is as follows:

http:// WRP500_ip_address /admin/reboot

Note The Upgrade/Resync/Reboot URL works only after the administrator logs in to the web GUI.

Configuration Profile

Because the WRP500 has two sets of parameters, one set for data and one set for voice, the requirements vary from the provisioning of other Cisco Small Business IP Telephony Devices. You will have two profiles: one for the data (router) parameters and one for the voice parameters. One benefit of having separate profiles for voice parameters and data parameters is that you can deploy the common data parameters to all of your customer sites and deploy the custom voice parameters to each site individually.

- Data (router) parameters: Use the XML format only, as described in the Provisioning Guide

- Voice parameters: Use the XML format. The binary format is generated by a profile compiler tool available from Cisco. Find the correct SPA Profiler Compiler (SPC) for the firmware that you have installed on your WRP500. For more information about the data parameters, see Appendix A, “Advanced Voice Fields.”

Note You can download the SPC tools at the following URL: http://www.cisco.com/c/en/us/products/unifiedcommunications/wrp500-wireless-ac-broadband-router-2-phone-ports/index.html

XML Format

Use the XML format for data (router) parameters. The XML file consists of a series of elements (one per configuration parameter), encapsulated within the element tags <flat-profile> … </flat-profile>. The encapsulated elements specify values for individual parameters. Here is an example of a valid XML profile:

<flat-profile>

<Web_Remote_Management>0</Web_Remote_Management>

<Web_Remote_Upgrade>0</Web_Remote_Upgrade>

</flat-profile>

The names of parameters in XML profiles can generally be inferred from the WRP500 Configuration Utility, by substituting underscores (_) for spaces and other control characters. To distinguish between Lines 1, 2, 3, and 4, corresponding parameter names are augmented by the strings _1_, _2_, _3_, and _4_. For example, Line 1 Proxy is named Proxy_1_ in XML profiles.

Binary Format

The WRP500 does not support binary format files.

Configure Your System for ITSP Interoperability

This chapter provides configuration details to help you to ensure that your infrastructure properly supports voice services.

- Configure NAT Mapping

- Firewalls and SIP

- Configure SIP Timer Values

Configure NAT Mapping

As discussed in “Product Overview and Deployment Guidelines,” some form of Network Address Translation (NAT) mapping is needed to support VoIP. If your ITSP does not support NAT mapping through a Session Border Controller, and if your edge device is not a SIP-ALG router, you can address this issue through one of the following methods:

- Configure NAT Mapping with a Static IP Address

- Configure NAT Mapping with STUN

Configure NAT Mapping with a Static IP Address

This option can be used if the following requirements are met:

- You must have a static external (public) IP address from your ISP.

- The edge device—that is, the router between your local area network and your ISP network—must have a symmetric NAT mechanism. If the WRP500 is the edge device, this requirement is met. If another device is used as the edge device, see the “Determine Whether the Router Uses Symmetric or Asymmetric NAT” section.

- If the WRP500 is connected to an Ethernet switch, the switch must be configured to enable Spanning Tree Protocol and Port Fast on the port to which the WRP500 is connected.

Note Use NAT mapping only if the ITSP network does not provide a Session Border Controller functionality.

Step 1 Log in as administrator.

Step 2 Under the Voice menu, click SIP.

Step 3 In the NAT Support Parameters section, enter the following settings:

- Substitute VIA Addr: Choose yes.

- EXT IP: Enter the public IP address that was assigned by your ISP.

Figure 2-1 Voice tab > SIP: NAT Support Parameters

Step 4 Under the Voice menu, click Line 1 or Line 2 to choose the line interface that you want to modify.

Step 5 In the NAT Settings section, enter the following settings:

- NAT Mapping Enable: Choose yes.

- NAT Keep Alive Enable: Choose yes.

Figure 2-2 Voice tab > Line N > NAT Settings

Step 6 Click Submit.

Note You also need to configure the firewall settings on your router to allow SIP traffic. See Firewalls and SIP.

Configure NAT Mapping with STUN

This option is considered a practice of last resort and should be used only if the other methods are unavailable. This option can be used if the following requirements are met:

- You have a dynamically assigned external (public) IP address from your ISP.

- You must have a computer running STUN server software.

- The edge device uses an asymmetric NAT mechanism. If the WRP500 is the edge device, this requirement is not met . For more information, see the “Determine Whether the Router Uses Symmetric or Asymmetric NAT” section.

- If the WRP500 is connected to an Ethernet switch, the switch must be configured to enable Spanning Tree Protocol and Port Fast on the port to which the WRP500 is connected.

Note Use NAT mapping only if the ITSP network does not provide a Session Border Controller functionality.

Step 1 Log in as administrator.

Step 2 Under the Voice menu, click SIP.

Step 3 In the NAT Support Parameters section, enter the following settings:

- Substitute VIA Addr: yes

- STUN Enable : Choose yes.

- STUN Test Enable: Choose yes.

- STUN Server: Enter the IP address for your STUN server.

Figure 2-3 Voice tab > SIP > NAT Support Parameters

Step 4 Under the Voice menu, click Line 1 or Line 2 to choose the line interface that you want to modify.

Step 5 In the NAT Settings section, enter the following settings:

- NAT Mapping Enable: Choose yes.

- NAT Keep Alive Enable: Choose yes (optional).

Figure 2-4 Voice tab > Line N > NAT Settings

Note Your ITSP may require the WRP500 to send NAT keep alive messages to keep the NAT ports open permanently. Check with your ITSP to determine the requirements.

Step 6 Click Submit.

Note You also need to configure the firewall settings on your router to allow SIP traffic. See the “Firewalls and SIP” section.

Determine Whether the Router Uses Symmetric or Asymmetric NAT

To use a STUN server, the edge device—that is, the device that routes traffic between your private network and your ISP network—must have an asymmetric NAT mechanism. You need to determine which type of NAT mechanism is available on that device.

STUN does not work on routers with symmetric NAT. With symmetric NAT, IP addresses are mapped from one internal IP address and port to one external, routable destination IP address and port. If another packet is sent from the same source IP address and port to a different destination, a different IP address and port number combination is used. This method is restrictive because an external host can send a packet to a particular port on the internal host only if the internal host first sent a packet from that port to the external host.

Note This procedure assumes that a syslog server is configured and is ready to receive syslog messages.

Step 1 Make sure that no firewall is running on your computer that could block the syslog port (port 514 by default).

Step 2 Log in as administrator.

Step 3 To enable debugging, complete the following tasks:

a. Under the Voice menu, click System .

b. In the Syslog Server and Debug Server fields, enter the IP address of your syslog server. This address and port number must be reachable from the WRP500.

c. From the Debug level drop-down list, choose 3.

d. From the Debug option drop-down list, choose dbg_all.

Figure 2-5 Voice tab > System

Step 4 To collect information about the type of NAT that your router is using, complete the following tasks:

a. Under the Voice menu, click SIP.

b. Scroll down to the NAT Support Parameters section.

c. From the STUN Test Enable field, choose yes.

Step 5 To enable SIP signaling, complete the following task:

a. Under the Voice menu, click Line 1 or Line 2 to choose the line interface that you want to modify.

b. In the SIP Settings section, choose full from the SIP Debug Option field.

Step 6 Click Submit .

Step 7 View the syslog messages to determine whether your network uses symmetric NAT. Look for a warning header in the REGISTER messages, such as Warning: 399 spa "Full Cone NAT Detected.”

Firewalls and SIP

To enable SIP requests and responses to be exchanged with the SIP proxy at the ITSP, you must ensure that your firewall allows both SIP and RTP unimpeded access to the Internet.

- Make sure that the following ports are not blocked:

– SIP ports—UDP port 5060 through 5061, which are used for the ITSP line interfaces

– RTP ports—16384 to 16482

- Also disable SPI (Stateful Packet Inspection) if this function exists on your firewall.

Configure SIP Timer Values

The default timer values should be adequate in most circumstances. However, you can adjust the SIP timer values as needed to ensure interoperability with your ITSP. For example, if SIP requests are returned with an “invalid certificate” message, you may need to enter a longer SIP T1 retry value.

For more information, see the “SIP Timer Values (sec) section” section.

Configure Voice Services

This chapter describes how to configure your WRP500 to meet customer requirements for voice services.

- Analog Telephone Adapter Operations

- ATA Software Features

- Register to the Service Provider

- Manage Caller ID Service

- Optimize Fax Completion Rates

- Silence Suppression and Comfort Noise Generation

- Configure Dial Plans

- Secure Call Implementation

Analog Telephone Adapter Operations

The WRP500 is equipped with a built-in Analog Telephone Adapter (ATA). An ATA is an intelligent low-density Voice over IP (VoIP) gateway that enables carrier-class residential and business IP Telephony services that are delivered over broadband or high-speed Internet connections. Users can access Internet phone services through standard analog telephone equipment.

The WRP500 maintains the state of each call it terminates and reacts properly to user input events (such as on/off hook or hook flash). The WRP500 uses the Session Initiation Protocol (SIP) open standard, so little or no involvement by a “middle-man” server or media gateway controller occurs. SIP allows interoperation with all Internet telephony service providers (ITSPs) that support SIP.

ATA Software Features

The WRP500 is equipped with a full featured, fully programmable ATA that can be custom provisioned within a wide range of configuration parameters. These sections describe the factors that contribute to voice quality:

- Supported Codecs

- SIP Proxy Redundancy

- Other ATA Software Features

Supported Codecs

The WRP500 supports the following codecs:

- G.711u (configured by default) and G.711a

G.711 (A-law and mu-law) are very low complexity codecs that support uncompressed 64 kbps digitized voice transmissions at one through ten 5-millisecond voice frames per packet. This codec provides the highest voice quality and uses the most bandwidth of any of the available codecs.

- G.729a

The ITU G.729 voice coding algorithm is used to compress digitized speech. G.729a is a reduced complexity version of G.729. It requires about half the processing power as compared to G.729. The G.729 and G.729a bit streams are compatible and interoperable, but not identical.

The administrator can select the preferred codecs to be used for each line. See the “Audio Configuration section” section.

In addition, negotiation of the optimal voice codec sometimes depends on the ability of an ATA to match a codec name with the codec that the far-end device uses. You can individually name the various codecs so that the WRP500 can successfully negotiate the codec with the far-end equipment. For more information, see the Audio Configuration section.

SIP Proxy Redundancy

In typical commercial IP Telephony deployments, all calls are established through a SIP proxy server. An average SIP proxy server may handle thousands of subscribers. It is important that a backup server be available so that an active server can be temporarily switched out for maintenance. The WRP500 supports the use of backup SIP proxy servers (via DNS SRV) so that service disruption should be nearly eliminated.

A relatively simple way to support proxy redundancy is to configure your DNS server with a list of SIP proxy addresses. The WRP500 can be instructed to contact a SIP proxy server in a domain named in the SIP message. The WRP500 consults the DNS server to get a list of hosts in the given domain that provides 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 WRP500 tries to contact the list of hosts in the order of their stated priority.

If the WRP500 is currently using a lower priority proxy server, it periodically probes the higher priority proxy to check whether it is back on line, and switches back to the higher priority proxy when possible. SIP Proxy Redundancy is configured in the Line and PSTN Line pages in the Configuration Utility.

Other ATA Software Features

Table 3-1 summarizes other features that the WRP500 provides.

Table 3-1 ATA Software Features

Feature

Description

Silence Suppression

See “Silence Suppression and Comfort Noise Generation” section .

Modem and Fax Pass-Through

- Modem pass-through mode can be triggered only by predialing the number that is set in the Modem Line Toggle Code. (Set in the Regional tab.)

- FAX pass-through mode is triggered by a CED/CNG tone or by an NSE event.

- Echo canceler is automatically disabled for Modem pass-through mode.

Adaptive Jitter Buffer

The WRP500 can buffer incoming voice packets to minimize out-of-order packet arrival. This process is known as jitter buffering. The jitter buffer size proactively adjusts or adapts in size, depending on changing network conditions.

The WRP500 has a Network Jitter Level control setting for each line of service. The jitter level determines how aggressively the WRP500 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.

Adaptive Jitter Buffer is configured in the Line and PSTN Line tabs. See Appendix A, “Advanced Voice Fields.”

International Caller ID Delivery

In addition to support of the Bellcore (FSK) and Swedish/Danish (DTMF) methods of Caller ID (CID) delivery, ATAs provide a large subset of ETSI-compliant methods to support international CID equipment. International CID is configured in the Line and PSTN Line tabs. See Appendix A, “Advanced Voice Fields.”

Secure Calls

A user (if enabled by service provider or administrator) has the option to make an outbound call secure in the sense that the audio packets in both directions are encrypted. See the “Secure Call Implementation” section .

Adjustable Audio Frames Per Packet

This feature allows the user to set the number of audio frames that are contained in one RTP packet. Packets can be adjusted to contain audio frames of 10ms to 30ms in length. Increasing the time of packets decreases the bandwidth utilized, but it also increases delay and may affect voice quality. See the RTP Packet Size parameter found in the SIP tab in Appendix A, “Advanced Voice Fields.”

DTMF

The WRP500 may relay DTMF digits as out-of-band events to preserve the fidelity of the digits. This can enhance the reliability of DTMF transmission that many IVR applications, such as dial-up banking and airline information, require. DTMF is configured in the DTMF Tx Mode parameter that is found in the Line tabs. See Appendix A, “Advanced Voice Fields.”

Call Progress Tone Generation

The WRP500 has configurable call progress tones. Call progress tones are generated locally on the WRP500 so an end user is advised of status (such as ringback). Parameters for each type of tone (for instance, a dial tone that is played back to an end user) may include frequency and amplitude of each component, and cadence information. See the Regional tab in Appendix A, “Advanced Voice Fields.”

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. See the Regional tab in Appendix A, “Advanced Voice Fields.”

Echo Cancellation

Impedance mismatch between the telephone and the IP Telephony gateway phone port can lead to near-end echo. The WRP500 has a near-end echo canceler that compensates for impedance match. The WRP500 also implements an echo suppressor with comfort noise generator (CNG) so that any residual echo is not noticeable. Echo Cancellation is configured in the Regional, Line, and PSTN Line tabs. See Appendix A, “Advanced Voice Fields.”

Signaling Hook Flash Event

The WRP500 can signal hook flash events to the remote party on a connected call. This feature can be used to provide advanced mid-call services with third-party-call-control. Depending on the features that the service provider offers using third-party-call-control, the following ATA features may be disabled to correctly signal a hook-flash event to the softswitch:

- Call Waiting Service (parameter call waiting serv set in the Line tab)

- Three Way Conference Service (parameter three-way conf serv set in the Line tab)

- Three Way Call Service (parameter three-way call serv set in the Line tab)

You can configure the length of time allowed for detection of a hook flash using the Hook Flash Timer parameter on the Regional tab of the Configuration Utility. See Appendix A, “Advanced Voice Fields.”

Configurable Dial Plan with Interdigit Timers

The WRP500 has three configurable interdigit timers:

- Initial timeout (T)—Signals that the handset is off the hook and that no digit has been pressed yet.

- Long timeout (L)—Signals the end of a dial string; that is, no more digits are expected.

- Short timeout (S)—Used between digits; that is, after a digit is pressed, a short timeout prevents the digit from being recognized a second time.

See “Configure Dial Plans” section for more information.

Polarity Control

The WRP500 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. Polarity Control is configured in the Line and PSTN Line tabs. See Appendix A, “Advanced Voice Fields.”

Calling Party Control

Calling Party Control (CPC) signals to the called party equipment that the calling party has hung up during a connected call by removing the voltage between the tip and ring momentarily. This feature is useful for auto-answer equipment, which then knows when to disengage. CPC is configured in the Regional, Line, and PSTN Line tabs. See Appendix A, “Advanced Voice Fields.”

Syslog and Debug Server Records

Syslog and Debug Sever Records log more details than Report Generation and Event Logging. Using the configuration parameters, the WRP500 allows you to select which type of activity/events should be logged. Syslog and Debug Server allow the information to be sent to a Syslog Server. Syslog and Debug Server Records are configured in the System, Line, and PSTN Line tabs. See Appendix A, “Advanced Voice Fields.”

SIP Over TLS

The WRP500 allows the use of SIP over Transport Layer Security (TLS). SIP over TLS is designed to eliminate the possibility of malicious activity by encrypting the SIP messages of the service provider and the end user. SIP over TLS relies on the widely deployed and standardized TLS protocol. SIP Over TLS encrypts only the signaling messages and not the media. A separate secure protocol, such as Secure Real-Time Transport Protocol (SRTP), can be used to encrypt voice packets. SIP over TLS is configured in the SIP Transport parameter configured in the Line tab(s). See Appendix A, “Advanced Voice Fields.”

Register to the Service Provider

To use VoIP phone service, you must configure your WRP500 to the Internet Telephony Service Provider (ITSP).

Note Each line tab must be configured separately. Each line tab can be configured for a different ITSP.

Step 1 Log in as administrator.

Step 2 Under the Voice menu, click Line 1 or Line 2 to choose the line interface that you want to modify.

Step 3 In the Proxy and Registration section, enter the Proxy.

Step 4 In the Subscriber Information section, enter the User ID and Password.

Note These are the minimum settings for most ITSP connections. Enter the account information as required by your ITSP.

Step 5 Click Submit. The devices reboot.

Step 6 To verify your progress, perform the following tasks:

- Under the Voice menu, click Info . Scroll down to the Line 1 Status or Line 2 Status section of the page, depending on which line you configured. Verify that the line is registered. Refer to the following example.

- Use an external phone to place an inbound call to the telephone number that was assigned by your ITSP. Assuming that you have left the default settings in place, the phone should ring and you can pick up the phone to get two-way audio.

- 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 succeed. Also verify that your DNS is configured properly.

Manage Caller ID Service

The choice of caller ID (CID) method is dependent on your area/region. To configure CID, use the following parameters:

Parameter

Tab

Description and Value

Caller ID Method

Regional

The following choices are available:

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

- DTMF (Finland, Sweden) —CID only. 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 CPE 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 CPE after DTAS for CIDCW. Polarity reversal is applied only if equipment is on hook.

The default is Bellcore (N.Amer, China) .

Caller ID FSK Standard

Regional

The WRP500 supports bell 202 and v.23 standards for caller ID generation. Select the FSK standard you want to use, bell 202 or v.23.

The default is bell 202.

Three types of Caller ID exist:

- On Hook Caller ID Associated with Ringing — This type of Caller ID is used for incoming calls when the attached phone is on hook. See the following figure (a) – (c). All CID methods can be applied for this type of CID.

- On Hook Caller ID Not Associated with Ringing — This feature is used to send VMWI signal to the phone to turn the message waiting light on and off. See the following figure (d) and (e). This is available only for FSK-based CID methods: Bellcore, ETSI FSK, and ETSI FSK With PR.

- Off Hook Caller ID — This is used to delivery caller-id on incoming calls when the attached phone is off hook. (See the following figure.) This can be call waiting caller ID (CIDCW) or to notify the user that the far-end party identity has changed or updated (such as due to a call transfer). This is available only for FSK-based CID methods: Bellcore, ETSI FSK, and ETSI FSK With PR.

Optimize Fax Completion Rates

Issues can occur with fax transmissions over IP networks, even with the T.38 standard, which is supported by the WRP500. You can adjust several settings on your WRP500 to optimize your fax completion rates.

Note Only T.38 Fax is supported. The WRP500 supports one connection.

Step 1 Ensure that you have enough bandwidth for the uplink and the downlink:

- For G.711 fallback, approximately 100 kbps are recommended.

- For T.38, allocate at least 50 kbps.

Step 2 To optimize G.711 fallback fax completion rates, set the following on the Line tab of your ATA device:

- Call Waiting Serv: no

- Three Way Call Serv: no

- Preferred Codec: G.711

- Use pref. codec only: yes

Step 3 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

Step 4 Enable T.38 fax on the WRP500 by configuring the following parameter on the Line tab for the FXS port to which the FAX machine is connected:

FAX_Enable T38: Yes

Note If a T.38 call cannot be set up, the call automatically reverts to G.711 fallback.

Step 5 If you are using a Cisco media gateway, use the following settings:

Make sure the Cisco gateway is correctly configured for T.38 with the SPA dial peer. For example:

fax protocol T38

fax rate voice

fax-relay ecm disable

fax nsf 000000

no vad

Fax Troubleshooting

If 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 following statistics:

- Jitter

- Loss

- Delay

Step 5 If faxes fail consistently, capture a copy of the voice settings by selecting Save As > Web page, complete from the administration web server page. You can send this configuration file to Technical Support.

Step 6 Enable and capture the debug log. For instructions, refer to Appendix C, “Troubleshooting.”

Note You may also capture data by using a sniffer trace.

Step 7 Identify the type of fax machine that is connected to the ATA device.

Step 8 Contact technical support:

– If you are an end user of VoIP products, contact the reseller or Internet telephony service provider (ITSP) that supplied the equipment.

– If you are an authorized Cisco partner, contact Cisco technical support.

Silence Suppression and Comfort Noise Generation

Voice Activity Detection (VAD) with Silence Suppression is a means of increasing the number of calls that the network supports by reducing the required bandwidth for a single call. VAD uses a sophisticated algorithm to distinguish between speech and non-speech signals. Based on the current and past statistics, the VAD algorithm decides whether speech is present. If the VAD algorithm decides speech is not present, silence suppression and comfort noise generation is activated. This is accomplished by removing and not transmitting the natural silence that occurs in a normal two-way connection. The IP bandwidth is used only when someone is speaking. During the silent periods of a telephone call, additional bandwidth is available for other voice calls or data traffic because the silence packets are not being transmitted across the network.

Comfort Noise Generation provides artificially-generated background white noise (sounds), designed to reassure callers that their calls are still connected during silent periods. If Comfort Noise Generation is not used, the caller may think the call has been disconnected because of the “dead silence” periods that the VAD and Silence Suppression feature creates.

Silence suppression is configured in the Line tab.

Configure Dial Plans

Dial plans determine how the digits are interpreted and transmitted. They also determine whether the dialed number is accepted or rejected. You can use a dial plan to facilitate dialing or to block certain types of calls, such as long distance or international.

This section includes information that you need to understand dial plans, as well as procedures for configuring your own dial plans. This section includes the following topics:

- About Dial Plans

- Edit Dial Plans

About Dial Plans

This section provides information to help you understand how dial plans are implemented.

Refer to the following topics:

- Digit Sequences

- Digit Sequence Examples

- Acceptance and Transmission of Dialed Digits

- Dial Plan Timer (Off-Hook Timer)

- Interdigit Long Timer (Incomplete Entry Timer)

- Interdigit Short Timer (Complete Entry Timer)

Digit Sequences

A dial plan contains a series of digit sequences, separated by the | character. The entire collection of sequences is enclosed within parentheses. Each digit sequence within the dial plan consists of a series of elements, which are individually matched to the keys that the user presses.

Note White space is ignored, but may be used for readability.

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, enter [2-9] to allow the user to press any one digit from 2 through 9.

- Numeric range with other characters

For example, 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 0 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 )

- Extensions on your system

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 )

[1-8]xx Allows a user to 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

EXAMPLE: ( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9, 011xxxxxx. | 0 | [49]111)

9, xxxxxxx After a user presses 9, an external dial tone sounds. The user can enter any seven-digit number, as in a local call.

- Local dialing with 3-digit area code and a 7-digit local number

EXAMPLE: ( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9, 011xxxxxx. | 0 | [49]11 )

9, <:1>[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

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 )

8, <:1212>xxxxxxx This 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

EXAMPLE: ( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9, 011xxxxxx. | 0 | [49]11 )

9, 1 [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

EXAMPLE: ( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9, 011xxxxxx. | 0 | [49]11 )

9, 1 900 xxxxxxx ! This digit sequence is useful if you want to prevent users from dialing numbers that are associated with high tolls or inappropriate content, such as 1-900 numbers in the U.S. After the user presses 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

EXAMPLE: ( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9, 011xxxxxx. | 0 | [49]11 )

9, 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 U.S.

- Informational numbers

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 )

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 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 the user enters more digits, the set of candidates diminishes until only one or none are valid. When a terminating event occurs, the WRP500 either accepts the user-dialed sequence and initiates a call, or 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. The default value is 10 seconds.

- The Interdigit Short Timer applies when the dialed digits match one or more candidate sequences in the dial plan. The default value is 3 seconds.

The user presses the # key or the dial softkey on the phone display.

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. The default value is 5.

Syntax for the Dial Plan Timer

SYNTAX: (P s <: 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 an extension number or a DID number. No wildcard characters are allowed because the number will be transmitted as shown. If you omit the number substitution, <:n>, the user hears a reorder (fast busy) tone after the specified number of seconds.

Examples for the Dial Plan Timer

- Allow more time for users to start dialing after taking a phone off hook.

EXAMPLE: ( 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.

- Create a hotline for all sequences on the System Dial Plan

EXAMPLE: ( 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

EXAMPLE: ( P0 <:1000>)

With the timer set to 0 seconds, the call is transmitted automatically to the specified extension when the phone goes off hook. Enter this sequence in the Phone Dial Plan for Ext 2 or higher on a client station.

Interdigit Long Timer (Incomplete Entry Timer)

You can think of this timer as the “incomplete entry” timer. This timer measures the interval between dialed digits. It applies as long as the dialed digits do not match any digit sequences in the dial plan. Unless the user enters another digit within the specified number of seconds, the entry is evaluated as incomplete, and the call is rejected. The default value is 10 seconds.

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 the “Reset the Control Timers” section.

Syntax for the Interdigit Long Timer

SYNTAX: L :s, ( dial plan )

- s: The number of seconds; if no number is entered after L: , the default timer of 5 seconds applies.

- Note that the timer sequence appears to the left of the initial parenthesis for the dial plan.

Example for the Interdigit Long Timer

EXAMPLE: 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. This setting is especially helpful to users such as sales people, who are reading the numbers from business cards and other printed materials while dialing.

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. The default value is 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.

EXAMPLE: 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. This setting is especially helpful to users such as sales people, who are reading the numbers from business cards and other printed materials while dialing.

- Set an instant timer for a particular sequence within the dial plan.

EXAMPLE: (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.

Edit Dial Plans

You can edit dial plans and can modify the control timers.

Enter the Line Interface Dial Plan

This dial plan is used to strip steering digits from a dialed number before it is transmitted out to the carrier.

Step 1 Start Internet Explorer, connect to the Configuration Utility, choose Voice . If prompted, enter the administrative login provided by the Service Provider. (The default username and password are both admin .)

Step 2 Under the Voice menu, click Line 1 or Line 2 , depending on the line interface that you want to configure.

Step 3 Scroll down to the Dial Plan section.

Step 4 Enter the digit sequences in the Dial Plan field. For more information, see the “About Dial Plans” section.

Step 5 Click Submit.

Reset the Control Timers

You can use the following procedure to reset the default timer settings for all calls.

Note If you need to edit a timer setting only for a particular digit sequence or type of call, you can edit the dial plan. See the “About Dial Plans” section.

Step 1 Start Internet Explorer, connect to the Configuration Utility, choose Voice . If prompted, enter the administrative login provided by the Service Provider. (The default username and password are both admin .)

Step 2 Under the Voice menu, click Regional.

Step 3 Scroll down to the Control Timer Values section.

Step 4 Enter the desired values in the Interdigit Long Timer field and the Interdigit Short Timer field. Refer to the definitions at the beginning of this section.

Secure Call Implementation

This section describes secure call implementation with the WRP500. It includes the following topics:

- Enable Secure Calls

Note This is an advanced topic meant for experience installers. Also see the Provisioning Guide at the following URL:

http://www.cisco.com/c/dam/en/us/td/docs/voice_ip_comm/csbpvga/ata/provisioning/guide/Provisioning.pdf

Enable Secure Calls

WRP500 does not support establishing secure call by “mini certificate” as WRP400 did. The only method to enable a secure call requires use of SRTP, while the SRTP key parameters are transferred in SIP messages that are encrypted by TLS.

To enable SRTP on Line 1:

- Voice > Line 1 > Secure Call Serv, set to Yes

- Voice > User 1 > Secure Call Setting, set to Yes

To enable SIP over TLS on Line:

- Voice > Line 1 > SIP Transport, set to TLS

Advanced Voice Fields

This appendix describes the Advanced settings that are available after you log in as administrator.

After you click the Voice tab, you can choose the following pages:

- Info page

- System page

- SIP page

- Regional page

- Line page

- User page

Info page

You can use the Voice tab > Info page to view information about the WRP500. This page includes the following sections:

- Product Information section

- System Status section

- Line Status section

Note The fields on the Info page are read-only and cannot be edited.

Product Information section

This table describes the fields in the Product Information section of the Voice tab > Info page.

Field

Description

Product Name

Model number/name.

Serial Number

Serial number.

Software Version

Software version number.

Hardware Version

Hardware version number.

MAC Address

MAC address.

Client Certificate

Status of the client certificate, which can indicate whether the WRP500 has been authorized by your ITSP.

Customization

For a Remote Configuration (RC) unit, this field indicates whether the unit has been customized or not. Pending indicates a new RC unit that is ready for provisioning. If the unit has already retrieved its customized profile, this field displays the name of the company that provisioned the unit.

Voice Module Version

Voice module number.

System Status section

This table describes the fields in the System Status section of the Voice tab > Info page.

Field

Description

Current Time

Current date and time of the system; for example, 10/3/2003 16:43:00.

Elapsed Time

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

RTP Packets Sent

Total number of RTP packets sent (including redundant packets).

RTP Bytes Sent

Total number of RTP bytes sent.

RTP Packets Recv

Total number of RTP packets received (including redundant packets).

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

External IP address used for NAT mapping.

Line Status section

This table describes the fields in the Line Status section of the Voice tab > Info page.

Field

Description

Hook State

Hook state of the FXS port. Options are either 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

Number of seconds before the next registration renewal.

Message Waiting

Indicates whether you have new voice mail waiting. Options are either Yes or No. The value automatically is set to Yes when a message is received. You also can clear or set the flag manually. Setting this value to Yes can activate stutter tone and VMWI signal. This parameter is stored in long term memory and survives after reboot or power cycle.

Call Back Active

Indicates whether a call back request is in progress. Options are either Yes or No.

Last Called Number

The last number called from the FXS line.

Last Caller Number

Number of the last caller.

Mapped SIP Port

Port number of the SIP port mapped by NAT.

Call 1 and 2 State

May take one of the following values:

- Idle

- Dialing

- Stunning

- Calling

- Proceeding

- Ringing

- Invalid

- Connected

- Hold

- Holding

- Resuming

- Transit

Call 1 and 2 Tone

Type of tone used by the call.

Call 1 and 2 Encoder

Codec used for encoding.

Call 1 and 2 Decoder

Codec used for decoding.

Call 1 and 2 FAX

Status of the fax mode.

Call 1 and 2 Type

Direction of the call. May take one of the following values:

- Inbound

- Outbound

- Transferred

Call 1 and 2 Remote Hold

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

Call 1 and 2 Callback

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

Call 1 and 2 Peer Name

Name of the internal phone.

Call 1 and 2 Peer Phone

Phone number of the internal phone.

Call 1 and 2 Call Duration

Duration of the call.

Call 1 and 2 Packets Sent

Number of packets sent.

Call 1 and 2 Packets Recv

Number of packets received.

Call 1 and 2 Bytes Sent

Number of bytes sent.

Call 1 and 2 Bytes Recv

Number of bytes received.

Call 1 and 2 Decode Latency

Number of milliseconds for decoder latency.

Call 1 and 2 Jitter

Number of milliseconds for receiver jitter.

Call 1 and 2 Packets Lost

Number of packets lost.

Call 1 and 2 Packet Error

Number of invalid packets received.

Call 1 and 2 Mapped RTP Port

The port mapped for Real Time Protocol traffic for Call 1/2.

Call 1 and 2 Media Loopback

Media loopback is used to quantitatively and qualitatively measure the voice quality that the end user experiences.

System page

You can use the Voice tab > System page to configure your system and network connections. This page includes the following sections:

- System Configuration section

- Miscellaneous Settings section

System Configuration section

This table describes the fields in the System Configuration section of the Voice tab > System page.

Field

Description

Restricted Access Domains

This feature is used when implementing software customization.

IVR Admin Passwd

Password for entering IVR menu.

Miscellaneous Settings section

This table describes the fields in the Miscellaneous section of the Voice tab > System page.

Field

Description

Syslog Server

Specifies the IP address of the syslog server.

Debug Server

Specifies the IP address of the debug server, which logs debug information. The level of detailed output depends on the debug level parameter setting.

Debug Level

Determines the level of debug information that is generated. Select 0 , 1 , 2 , or 3 from the drop-down menu. The higher the debug level, the more debug information is generated.

The default is 0 , which indicates that no debug information is generated.

Debug Option

Specifies what debug information is expected. Generally can be set to dbg_all.

SIP page

You can use the Voice tab > SIP page to configure the SIP settings. This page includes the following sections:

- SIP Parameters section

- SIP Timer Values (sec) section

- Response Status Code Handling section

- RTP Parameters section

- SDP Payload Types section

- NAT Support Parameters section

SIP Parameters section

This table describes the fields in the SIP Parameters section of the Voice tab > SIP page.

Field

Description

Max Forward

SIP Max Forward value, which can range from 1 to 255.

The default is 70.

Max Redirection

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

The default is 5.

Max Auth

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

The default is 2.

SIP User Agent Name

User-Agent header used in outbound requests.

The default is $VERSION . If empty, the header is not included. Macro expansion of $A to $D corresponding to GPP_A to GPP_D allowed.

SIP Server Name

Server header used in responses to inbound responses.

The default is $VERSION .

SIP Reg User Agent Name

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.

The default is blank.

SIP Accept Language

Accept-Language header used. There is no default (this indicates the WRP500 does not include this header). If empty, the header is not included.

DTMF Relay MIME Type

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

The default is application/dtmf-relay.

Remove Last Reg

Lets you remove the last registration before registering a new one if the value is different. Select yes or no from the drop-down menu.

The default is no.

Use Compact Header

Lets you use compact SIP headers in outbound SIP messages. Select yes or no from the drop-down menu. If set to yes, the WRP500 uses compact SIP headers in outbound SIP messages. If set to no, the WRP500 uses normal SIP headers. If inbound SIP requests contain compact headers, the WRP500 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 WRP500 substitutes those headers with compact headers (if defined by RFC 261) if Use Compact Header parameter is set to yes.

The default is no.

Escape Display Name

Lets you keep the Display Name private. Select yes if you want the WRP500 to enclose the string (configured in the Display Name) in a pair of double quotes for outbound SIP messages. Any occurrences of or \ in the string is escaped with \ and \\ inside the pair of double quotes. Otherwise, select no.

The default is no.

RFC 2543 Call Hold

Configures the type of call hold: a:sendonly or 0.0.0.0.

The default is no; do not use the 0.0.0.0 syntax in a HOLD SDP; use the a:sendonly syntax.

Mark All AVT Packets

If set to yes, all AVT tone packets (encoded for redundancy) have the marker bit set. If set to no, only the first packet has the marker bit set for each DTMF event.

The default is yes.

SIP TCP Port Min

Specifies the lowest TCP port number that can be used for SIP sessions. The default Port Min is 5060.

SIP TCP Port Max

Specifies the highest TCP port number that can be used for SIP sessions. The default Port Max is 5080.

SIP Timer Values (sec) section

This table describes the fields in the SIP Timer Values section of the Voice tab > SIP page.

Field

Description

SIP T1

RFC 3261 T1 value (RTT estimate), which can range from 0 to 64 seconds.

The default is 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.

The default is 4.

SIP T4

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

The default is 5.

SIP Timer B

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

The default is 32 .

SIP Timer F

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

The default is 32 .

SIP Timer H

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

The default is 32 .

SIP Timer D

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

The default is 32 .

SIP Timer J

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

The default is 32 .

INVITE Expires

INVITE request Expires header value. If you enter 0, the Expires header is not included in the request.

The default is 240 . Range: 0–(2 31 –1).

ReINVITE Expires

ReINVITE request Expires header value. If you enter 0, the Expires header is not included in the request.

The default is 30 . Range: 0–(2 31 –1).

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.

The default is 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.

The default is 7200.

Reg Retry Intvl

Interval to wait before the WRP500 retries registration after failing during the last registration.

The default is 30.

Reg Retry Long Intvl

When registration fails with a SIP response code that does not match

Retry Reg RSC, the WRP500 waits for the specified length of time before retrying. If this interval is 0, the WRP500 stops trying. This value should be much larger than the Reg Retry Intvl value, which should not be 0.

The default is 1200 .

Response Status Code Handling section

This table describes the fields in the Response Status Code Handling section of the Voice tab > SIP page.

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.

SIT2 RSC

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

SIT3 RSC

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

SIT4 RSC

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

Try Backup RSC

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

Retry Reg RSC

Interval to wait before the WRP500 retries registration after failing during the last registration.

The default is 30.

RTP Parameters section

This table describes the fields in the RTP Parameters section of the Voice tab > SIP page.

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.

The default is 16384.

RTP Port Max

Maximum port number for RTP transmission and reception.

The default is 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.

The default is 0.030 .

Stats In BYE

Determines whether the WRP500 includes the P-RTP-Stat header or response to a BYE message. The header contains the RTP statistics of the current call. Select yes or no from the drop-down menu. 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 in s>,EN=<encoder>,DE=<decoder>.

The default is no.

SDP Payload Types section

This table describes the fields in the SDP Payload Types section of the Voice tab > SIP page.

Field

Description

NSE Dynamic Payload

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

The default is 100 .

AVT Dynamic Payload

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

The default is 101 .

INFOREQ Dynamic Payload

INFOREQ dynamic payload type.

There is no default.

NSE Codec Name

NSE codec name used in SDP.

The default is NSE .

AVT Codec Name

AVT codec name used in SDP.

The default is telephone-event .

G711u Codec Name

G.711u codec name used in SDP.

The default is PCMU .

G711a Codec Name

G.711a codec name used in SDP.

The default is PCMA .

G729a Codec Name

G.729a codec name used in SDP.

The default is G729a.

G729b Codec Name

G.729b codec name used in SDP.

The default is G729ab.

EncapRTP Codec Name

EncapRTP codec name used in SDP.

The default is EncapRTP.

EncapRTP Dynamic Payload

EncapRTP dynamic payload type.

NAT Support Parameters section

This table describes the fields in the NAT Support Parameters section of the Voice tab > SIP page.

Field

Description

Handle VIA received

If you select yes, the WRP500 processes the received parameter in the VIA header (this value is inserted by the server in a response to anyone of its requests). If you select no, the parameter is ignored. Select yes or no from the drop-down menu.

The default is no.

Handle VIA rport

If you select yes, the WRP500 processes the rport parameter in the VIA header (this value is inserted by the server in a response to anyone of its requests). If you select no, the parameter is ignored. Select yes or no from the drop-down menu.

The default is 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.

The default is 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.

The default is no.

Substitute VIA Addr

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

The default is no.

Send Resp To Src Port

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

The default is no.

STUN Enable

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

The default is no.

STUN Test Enable

If the STUN Enable feature is enabled and a valid STUN server is available, the WRP500 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 WRP500 detects symmetric NAT or a symmetric firewall, NAT mapping is disabled.

The default is no.

STUN Server

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

EXT IP

External IP address to substitute for the actual IP address of the WRP500 in all outgoing SIP messages. If 0.0.0.0 is specified, no IP address substitution is performed.

If this parameter is specified, the WRP500 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.

Note 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 WRP500 is the edge device, the second requirement is met.
The default is 0.0.0.0 .

EXT RTP Port Min

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

The default is 0.

NAT Keep Alive Intvl

Interval between NAT-mapping keep alive messages.

The default is 15.

Regional page

You can use the Voice tab > Regional page to localize your system with the appropriate regional settings. This page includes the following sections:

- Call Progress Tones section

- Distinctive Ring Patterns section

- Distinctive Call Waiting Tone Patterns section

- Distinctive Ring/CWT Pattern Names section

- Control Timer Values (sec) section

- Vertical Service Activation Codes section

- Outbound Call Codec Selection Codes section

- Miscellaneous section

Call Progress Tones section

This table describes the fields in the Call Progress Tones section of the Voice tab > Regional page.

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.

The default is 350@-19,440@-19;10(*/0/1+2) .

Second Dial Tone

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

The default is 420@-19,520@-19;10(*/0/1+2) .

Outside Dial Tone

Alternative to the Dial Tone. It prompts the user to enter an external phone number, as opposed to an internal extension. It is triggered by a, (comma) character encountered in the dial plan.

The default is 420@-19;10(*/0/1) .

Prompt Tone

Prompts the user to enter a call forwarding phone number.

The default is 520@-19,620@-19;10(*/0/1+2) .

Busy Tone

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

The default is 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.

The default is 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 Reorder Tone times out.

The default is 480@10,620@0;10(.125/.125/1+2)

Ring Back Tone

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

The default is 440@-19,480@-19;*(2/4/1+2) .

Ring Back 2 Tone

Your WRP500 plays this ringback tone instead of Ring Back Tone if the called party replies with a SIP 182 response without SDP to its outbound INVITE request. The default value is the same as Ring Back Tone, except the cadence is 1s on and 1s off.

The default is 440@-19,480@-19;*(1/1/1+2) .

Confirm Tone

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

The default is 600@-16; 1(.25/.25/1) .

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.

The default is 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.

The default is 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.

The default is 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.

The default is 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.

The default is 350@-19,440@-19;2(.1/.1/1+2);10(*/0/1+2) .

Cfwd Dial Tone

Played when all calls are forwarded.

The default is 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.

The default is 600@-19*(.1/.1/1,.1/.1/1,.1/9.5/1) .

Conference Tone

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

The default is 350@-19;20(.1/.1/1,.1/9.7/1) .

Secure Call Indication Tone

Played when a call has been successfully switched to secure mode. 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.

The default is 397@-19,507@-19;15(0/2/0,.2/.1/1,.1/2.1/2) .

Feature Invocation Tone

Played when a feature is implemented.

The default is 350@-16;*(.1/.1/1) .

Distinctive Ring Patterns section

This table describes the fields in the Distinctive Ring Patterns section of the Voice tab > Regional page.

Field

Description

Ring1 Cadence

Cadence script for distinctive ring 1.

The default is 60(2/4).

Ring2 Cadence

Cadence script for distinctive ring 2.

The default is 60(.8/.4,.8/4).

Ring3 Cadence

Cadence script for distinctive ring 3.

The default is 60(.4/.2,.4/.2,.8/4).

Ring4 Cadence

Cadence script for distinctive ring 4.

The default is 60(.3/.2,1/.2,.3/4).

Ring5 Cadence

Cadence script for distinctive ring 5.

The default is 1(.5/.5).

Ring6 Cadence

Cadence script for distinctive ring 6.

The default is 60(.2/.4,.2/.4,.2/4).

Ring7 Cadence

Cadence script for distinctive ring 7.

The default is 60(.4/.2,.4/.2,.4/4).

Ring8 Cadence

Cadence script for distinctive ring 8.

The default is 60(0.25/9.75).

Distinctive Call Waiting Tone Patterns section

This table describes the fields in the Distinctive Call Waiting Tone Patterns section of the Voice tab > Regional page.

Field

Description

CWT1 Cadence

Cadence script for distinctive CWT 1.

The default is 30(.3/9.7).

CWT2 Cadence

Cadence script for distinctive CWT 2.

The default is 30(.1/.1,.1/9.7).

CWT3 Cadence

Cadence script for distinctive CWT 3.

The default is 30(.1/.1,.1/.1,.1/9.7).

CWT4 Cadence

Cadence script for distinctive CWT 4.

The default is 30(.1/.1,.3/.1,.1/9.3).

CWT5 Cadence

Cadence script for distinctive CWT 5.

The default is 1(.5/.5).

CWT6 Cadence

Cadence script for distinctive CWT 6.

The default is 30(.1/.1,.3/.2,.3/9.1).

CWT7 Cadence

Cadence script for distinctive CWT 7.

The default is 30(.3/.1,.3/.1,.1/9.1).

CWT8 Cadence

Cadence script for distinctive CWT 8.

The default is 2.3(.3/2).

Distinctive Ring/CWT Pattern Names section

This table describes the fields in the Distinctive Ring/CWT Pattern Names section of the Voice tab > Regional page.

Field

Description

Ring1 Name

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

The default is Bellcore-r1.

Ring2 Name

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

The default is Bellcore-r2.

Ring3 Name

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

The default is Bellcore-r3.

Ring4 Name

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

The default is Bellcore-r4.

Ring5 Name

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

The default is Bellcore-r5.

Ring6 Name

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

The default is Bellcore-r6.

Ring7 Name

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

The default is Bellcore-r7.

Ring8 Name

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

The default is Bellcore-r8.

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 does not sound right, or your phone does not ring, change your Ring Waveform, Ring Frequency, and Ring Voltage to the following:

– Ring Waveform: Sinusoid

– Ring Frequency: 25

– Ring Voltage: 80V

Field

Description

Ring Waveform

Waveform for the ringing signal. Choices are Sinusoid or Trapezoid . The default is Trapezoid .

Ring Frequency

Frequency of the ringing signal. Valid values are 10–100 (Hz). The default is 20 .

Ring Voltage

Ringing voltage. Choices are 60–90 (V). The default is 85 .

CWT Frequency

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

The default is 440@-10 .

Control Timer Values (sec) section

This table describes the fields in the Control Timer Values (sec) section of the Voice tab > Regional page.

Field

Description

Hook Flash Timer Min

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

The default is 0.1 .

Hook Flash Timer Max

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

The default is 0.9 .

Callee On Hook Delay

Phone must be on-hook for at least this length of time in sec before the WRP500 tears down the current inbound call. This does not apply to outbound calls. Range: 0–255 seconds.

The default is 0 .

Reorder Delay

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

The default is 5 .

Call Back Expires

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

The default is 1800 .

Call Back Retry Intvl

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

The default is 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 WRP500 still considers the call as failed and keeps on retrying.

The default is 0.5 .

VMWI Refresh Intvl

Interval between VMWI refresh to the CPE.

The default is 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.

The default is 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.

The default is 3 .

CPC Delay

Delay in seconds after caller hangs up when the WRP500 starts removing the tip-and-ring voltage to the attached equipment of the called party. Range: 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.

The default is 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 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.

The default is 0 (CPC disabled).

Vertical Service Activation Codes section

Vertical Service Activation Codes are automatically appended to the dial plan. There is no need to include them in the dial plan, but no harm is done if they are included.

This table describes the fields in the Vertical Service Activation Codes section of the Voice tab > Regional page.

Field

Description

Call Return Code

This code calls the last caller.

The default is *69 .

Call Redial Code

Redials the last number called.

The default is *07 .

Blind Transfer Code

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

The default is *98 .

Call Back Act Code

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

The default is *66 .

Call Back Deact Code

Cancels a callback.

The default is *86 .

Call Back Busy Act Code

Starts a callback when the last outbound call is busy.

The default is *05

Cfwd All Act Code

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

The default is *72 .

Cfwd All Deact Code

Cancels call forwarding of all calls.

The default is *73 .

Cfwd Busy Act Code

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

The default is *90 .

Cfwd Busy Deact Code

Cancels call forwarding of busy calls.

The default is *91 .

Cfwd No Ans Act Code

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

The default is *92 .

Cfwd No Ans Deact Code

Cancels call forwarding of no-answer calls.

The default is *93 .

Cfwd Last Act Code

Forwards the last inbound or outbound calls to the extension specified after the activation code.

The default is *63 .

Cfwd Last Deact Code

Cancels call forwarding of the last inbound or outbound calls.

The default is *83 .

Block Last Act Code

Blocks the last inbound call.

The default is *60 .

Block Last Deact Code

Cancels blocking of the last inbound call.

The default is *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.

The default is *64 .

Accept Last Deact Code

Cancels the code to accept the last outbound call.

The default is *84 .

CW Act Code

Enables call waiting on all calls.

The default is *56 .

CW Deact Code

Disables call waiting on all calls.

The default is *57 .

CW Per Call Act Code

Enables call waiting for the next call.

The default is *71 .

CW Per Call Deact Code

Disables call waiting for the next call.

The default is *70 .

Block CID Act Code

Blocks caller ID on all outbound calls.

The default is *67 .

Block CID Deact Code

Removes caller ID blocking on all outbound calls.

The default is *68 .

Block CID Per Call Act Code

Blocks caller ID on the next outbound call.

The default is *81 .

Block CID Per Call Deact Code

Removes caller ID blocking on the next inbound call.

The default is *82 .

Block ANC Act Code

Blocks all anonymous calls.

The default is *77 .

Block ANC Deact Code

Removes blocking of all anonymous calls.

The default is *87 .

DND Act Code

Enables the do not disturb feature.

The default is *78 .

DND Deact Code

Disables the do not disturb feature.

The default is *79 .

CID Act Code

Enables caller ID generation.

The default is *65 .

CID Deact Code

Disables caller ID generation.

The default is *85 .

CWCID Act Code

Enables call waiting, caller ID generation.

The default is *25 .

CWCID Deact Code

Disables call waiting, caller ID generation.

The default is *45 .

Dist Ring Act Code

Enables the distinctive ringing feature.

The default is *26

Dist Ring Deact Code

Disables the distinctive ringing feature.

The default is *46 .

Speed Dial Act Code

Assigns a speed dial number.

The default is *74 .

Secure All Call Act Code

Makes all outbound calls secure.

The default is *16 .

Secure No Call Act Code

Makes all outbound calls not secure.

The default is *17 .

Secure One Call Act Code

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

The default is *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.)

The default is *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.

Attn-Xfer Act Code

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

Modem Line Toggle Code

Toggles the line to a modem.

The default is *99 . Modem pass-through mode can be triggered only by pre-dialing this code.

FAX Line Toggle Code

Toggles the line to a fax machine.

The default is #99 .

Referral Services Codes

These codes tell the WRP500 what to do when the user places the current call on hold and is listening to the second dial tone.

One or more *code can be configured into this parameter, such as *98, or *97|*98|*123, etc. Max total length is 79 chars. This parameter applies when the user places the current call on hold (by Hook Flash) and is listening to second dial tone. Each *code (and the following valid target number according to current dial plan) entered on the second dial-tone triggers the WRP500 to perform a blind transfer to a target number that is preceded by the service *code.

For example, after the user dials *98, the WRP500 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 WRP500 sends a blind REFER to the holding party with the Refer-To target equals to *98 target_number . This feature allows the WRP500 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 WRP500. You can empty the corresponding *code that you do not want the WRP500 to process.

Feature Dial Services Codes

These codes tell the WRP500 what to do when the user is listening to the first or second dial tone.

One or more *code can be configured into this parameter, such as *72, or *72|*74|*67|*82, etc. Max total length is 79 chars. This parameter applies when the user has a dial tone (first or second dial tone). Enter *code (and the following target number according to current dial plan) entered at the dial tone triggers the WRP500 to call the target number preceded by the *code. For example, after user dials *72, the WRP500 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 WRP500 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 WRP500. You can empty the corresponding *code that you do not want to the WRP500 to process.

You can add a parameter to each *code in Features Dial Services Codes to indicate what tone to play after the *code is entered, such as *72‘c‘|*67‘p‘. Below are a list of allowed tone parameters (note the use of back quotes surrounding the parameter w/o 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 WRP500 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 it in this parameter. In that case, simple add that *code in the dial plan and the WRP500 send INVITE *73@..... as usual when user dials *73.

Outbound Call Codec Selection Codes section

These codes are automatically appended to the dial plan. Thus, they do not need to be included in the dial plan, but there is no harm in doing so.

This table describes the fields in the Outbound Call Codec Section Codes section of the Voice tab > Regional page.

Field

Description

Prefer G711u Code

Makes this codec the preferred codec for the associated call.

The default is *017110 .

Force G711u Code

Makes this codec the only codec that can be used for the associated call.

The default is *027110 .

Prefer G711a Code

Makes this codec the preferred codec for the associated call.

The default is *017111

Force G711a Code

Makes this codec the only codec that can be used for the associated call.

The default is *027111 .

Prefer G729a Code

Makes this codec the preferred codec for the associated call.

The default is *01729 .

Force G729a Code

Makes this codec the only codec that can be used for the associated call.

The default is *02729 .

Miscellaneous section

This table describes the fields in the Miscellaneous section of the Voice tab > Regional page.

Field

Description

Set Local Date (mm/dd)

Sets the local date (mm stands for months and dd stands for days). The year is optional and uses two or four digits.

Set Local Time (HH/mm)

Sets the local time (hh stands for hours and mm stands for minutes). Seconds are optional.

FXS Port Impedance

Sets the electrical impedance of the FXS port. Choices are 600 , 900 , 600+2.16uF , 900+2.16uF , 270+750||150nF , 220+850||120nF , 220+820||115nF , or 200+600||100nF .

The default is 600 .

FXS Port Input Gain

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

The default is -3 .

FXS Port Output Gain

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

The default is -3 .

DTMF Playback Level

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

The default is -7.3 .

DTMF Playback Length

Local DTMF playback duration in milliseconds.

The default is .1 .

DTMF Playback Twist

Local DTMF playback duration.

The default is 1.3.

Caller ID Method

The following choices are available:

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

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

- DTMF (Denmark) —CID only. DTMF 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 CPE 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 CPE after DTAS for CIDCW. Polarity reversal is applied only if equipment is on hook.

The default is Bellcore(N.Amer, China) .

Caller ID FSK Standard

The WRP500 supports bell 202 and v.23 standards for caller ID generation. Select the FSK standard you want to use, bell 202 or v.23 .

The default is bell 202 .

Feature Invocation Method

Select the method you want to use, Default or Sweden default . The default is Default .

Line page

You can use the Voice tab > Line page to configure the lines for voice service. This page includes the following sections:

- Line Enable section

- Streaming Audio Server (SAS) section

- NAT Settings section

- Network Settings section

- SIP Settings section

- Call Feature Settings section

- Proxy and Registration section

- Subscriber Information section

- Supplementary Service Subscription section

- Audio Configuration section

- Dial Plan section

- FXS Port Polarity Configuration section

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

Line Enable section

This table describes the fields in the Line Enable section of the Voice tab > Line page.

Field

Description

Line Enable

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

The default is yes .

Streaming Audio Server (SAS) section

This table describes the fields in the Streaming Audio Server (SAS) section of the Voice tab > Line page.

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.

The default is 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 WRP500 ends this call with a SIP BYE message. The range is 0 to 255 seconds (0 means that the session refresh is disabled).

The default is 30 .

SAS Inbound RTP Sink

This setting works around devices that do not play inbound RTP if the streaming audio server line declares itself as a send-only device and tells the client not to stream out audio. Enter a Fully Qualified Domain Name (FQDN) or IP address of an RTP sink; this value is used by the streaming audio server line in the SDP of its 200 response to an inbound INVITE message from a client.

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 a FQDN or IP address of a 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 and, if specified, in the m = line of the SDP. If this value is not specified or 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.

The default value is empty.

NAT Settings section

This table describes the fields in the NAT Settings section of the Voice tab > Line page.

Field

Description

NAT Mapping Enable

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

The default is no .

NAT Keep Alive Enable

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

The default is no .

NAT Keep Alive Msg

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

The default is $ 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.

The default is $ PROXY .

Network Settings section

This table describes the fields in the Network Settings section of the Voice tab > Line page.

Field

Description

SIP ToS/DiffServ Value

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

The default is 0x68 .

SIP CoS Value [0-7]

CoS value for SIP messages.

The default is 3 .

RTP ToS/DiffServ Value

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

The default is 0xb8 .

RTP CoS Value [0-7]

CoS value for RTP data.

The default is 6 .

Network Jitter Min/Max

Determines how jitter buffer range of WRP500 when Network Jitter Mode is adaptive. Jitter buffer size is adjusted dynamically.

The default value of Network Jitter Min is 10ms .

The default value of Network Jitter Max is 200ms .

Network Jitter Mode

Specify whether the jitter buffer should be adjusted or use some constant interval value. Select the appropriate setting: adaptive , static .

The default is adaptive .

SIP Settings section

This table describes the fields in the SIP Settings section of the Voice tab > Line page.

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 . The default is UDP .

SIP Port

Port number of the SIP message listening and transmission port.

The default is 5060 .

SIP 100REL Enable

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

The default is no .

EXT SIP Port

The external SIP port number.

Auth Resync-Reboot

If this feature is enabled, the WRP500 authenticates the sender when it receives the NOTIFY resync reboot (RFC 2617) message. To use this feature, select yes. Otherwise, select no.

The default is 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.

SIP Remote-Party-ID

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

The default is yes .

SIP GUID

The Global Unique ID is generated for each line for each device. When it is enabled, the WRP500 adds a GUID header in the SIP request. The GUID is generated the first time the unit boots up and stays with the unit through rebooting and even factory reset. This feature was requested by Bell Canada (Nortel) to limit the registration of SIP accounts.

The default is no .

SIP Debug Option

SIP messages are received at or sent from the proxy listen port. This feature controls which SIP messages to log. Choices are as follows:

- none —No logging.

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

- 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.

The default is none.

RTP Log Intvl

The interval for the RTP log. The default value is 0.

Restrict Source IP

If Lines 1 and 2 use the same SIP Port value and the Restrict Source IP feature is enabled, the proxy IP address for Lines 1 and 2 is treated as an acceptable IP address for both lines. To enable the Restrict Source IP feature, select yes. Otherwise, select no. If configured, the WRP500 will drop all packets sent to its SIP Ports originated 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).

The default is no .

Referor Bye Delay

Controls when the WRP500 sends BYE to terminate stale call legs upon completion of call transfers. Multiple delay settings (Referor, Refer Target, Referee, and Refer-To Target) are configured on this screen. For the Referor Bye Delay, enter the appropriate period of time in seconds.

The default is 4 .

Refer Target Bye Delay

For the Refer Target Bye Delay, enter the appropriate period of time in seconds.

The default is 0 .

Referee Bye Delay

For the Referee Bye Delay, enter the appropriate period of time in seconds.

The default is 0 .

Refer-To Target Contact

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

The default is no .

Sticky 183

If this feature is enabled, the IP telephony 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.

The default is no .

Auth INVITE

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

Use Anonymous With RPID

Set value of Remote Party ID to “anonymous, yes”

Use Local Addr in FROM

Use IP address in From header, no

Reply 182 On Call Waiting

Send 182 response when enter call waiting, no

Call Feature Settings section

This table describes the fields in the Call Feature Settings section of the Voice tab > Line page.

Field

Description

Blind Attn-Xfer Enable

Enables the WRP500 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 WRP500 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.

The default is no .

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.

The default is yes .

MoH server

Address of music on hold server

Conference Bridge URL

URL of Conference server

Proxy and Registration section

This table describes the fields in the Proxy and Registration section of the Voice tab > Line page.

Field

Description

Proxy

SIP proxy server for all outbound requests.

Outbound Proxy

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

Use Outbound Proxy

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

The default is no .

Use OB Proxy In Dialog

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

The default is yes .

Register

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

The default is 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.

The default is no .

Register Expires

Allow answering inbound calls without successful (dynamic) registration by the unit. If proxy responded to REGISTER with a smaller Expires value, the WRP500 will renew registration based on this smaller value instead of the configured value. If registration failed with an Expires too brief error response, the WRP500 will retry with the value given in the Min-Expires header in the error response.

The default is 3600 .

Ans Call Without Reg

Expires value in sec in a REGISTER request. The WRP500 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

Use DNS SRV

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

The default is no .

DNS SRV Auto Prefix

If enabled, the WRP500 will automatically prefix the Proxy or Outbound Proxy name with _sip._udp when performing a DNS SRV lookup on that name.

The default is no .

Proxy Fallback Intvl

This parameter sets the delay (sec) after which the WRP500 will retry from the highest priority proxy (or outbound proxy) servers after it has failed over to a lower priority server. This parameter is useful only if the primary and backup proxy server list is provided to the WRP500 via DNS SRV record lookup on the server name. (Using multiple DNS A record per server name does not allow the notion of priority and so all hosts will be considered at the same priority and the WRP500 will not attempt to fall back after a fail over).

The default is 3600

Proxy Redundancy Method

The WRP500 will make an internal list of proxies returned in DNS SRV records. In normal mode, this list will contain proxies ranked by weight and priority.

if Based on SRV port is configured the WRP500 does normal first, and also inspect the port number based on 1st proxy’s port on the list.

The default is Normal .

Voice Mail Server

Enter the URL or IP address of the server.

Mailbox Subscribe Expires

Expiry time to the voice mail server. The time to send another subscribe message to the voice mail server. The default is 2147483647.

Subscriber Information section

This table describes the fields in the Subscriber Information section of the Voice tab > Line page.

Field

Description

Display Name

Display name for caller ID.

User ID

Extension number for this line.

Password

Password for this line.

Use Auth ID

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

The default is no .

Auth ID

Authentication ID for SIP authentication.

Directory Number

Enter the number for this line.

Supplementary Service Subscription section

The WRP500 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 WRP500.

This table describes the fields in the Supplementary Service Subscription section of the Voice tab > Line page.

Field

Description

Call Waiting Serv

Enable Call Waiting Service.

The default is yes .

Block CID Serv

Enable Block Caller ID Service.

The default is yes .

Block ANC Serv

Enable Block Anonymous Calls Service

The default is yes .

Dist Ring Serv

Enable Distinctive Ringing Service

The default is yes .

Cfwd All Serv

Enable Call Forward All Service

The default is yes .

Cfwd Busy Serv

Enable Call Forward Busy Service

The default is yes .

Cfwd No Ans Serv

Enable Call Forward No Answer Service

The default is yes .

Cfwd Sel Serv

Enable Call Forward Selective Service

The default is yes .

Cfwd Last Serv

Enable Forward Last Call Service

The default is yes .

Block Last Serv

Enable Block Last Call Service

The default is yes .

Accept Last Serv

Enable Accept Last Call Service

The default is yes .

DND Serv

Enable Do Not Disturb Service

The default is yes .

CID_Serv

Enable Caller ID Service

The default is yes .

CWCID Serv

Enable Call Waiting Caller ID Service

The default is yes .

Call Return Serv

Enable Call Return Service

The default is yes .

Call Redial Serv

Enable Call Redial Service.

Call Back Serv

Enable Call Back Service.

Three Way Call Serv

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

The default is yes .

Three Way Conf Serv

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

The default is yes .

Attn Transfer Serv

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

The default is yes .

Unattn Transfer Serv

Enable Unattended (Blind) Call Transfer Service.

The default is yes .

MWI Serv

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

The default is yes .

VMWI Serv

Enable VMWI Service (FSK).

The default is yes .

Speed Dial Serv

Enable Speed Dial Service.

The default is yes .

Secure Call Serv

Enable Secure Call Service.

The default is yes .

Referral Serv

Enable Referral Service. See the Referral Services Codes parameter for more details.

The default is yes .

Feature Dial Serv

Enable Feature Dial Service. See the Feature Dial Services Codes parameter for more details.

The default is yes .

Service Announcement Serv

Enable Service Announcement Service.

The default is no .

Audio Configuration section

A codec resource is considered as allocated if it has been included in the SDP codec list of an active call, even though it eventually may not be the one chosen for the connection. So, if the G.729a codec is enabled and included in the codec list, that resource is tied up until the end of the call whether or not the call actually uses G.729a. If the G.729a resource is already allocated and since only one G.729a resource is allowed per device, no other low-bit-rate codec may be allocated for subsequent calls; the only choices are G711a and G711u.

This table describes the fields in the Audio Configuration section of the Voice tab > Line page.

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 , G729a.

The default is G711u.

Second Preferred Codec

Second 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: Unspecified, G711u, G711a, G729a.

The default is Unspecified.

Third Preferred Codec

Third 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: Unspecified, G711u, G711a, G729a.

The default is Unspecifie d.

Use Pref Codec Only

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.

The default is no.

Silence Supp Enable

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

The default is no.

G729a Enable

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

The default is yes.

Echo Canc Enable

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

The default is yes.

Echo Supp Enable

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

The default is yes.

FAX CED Detect Enable

To enable detection of the fax Caller-Entered Digits (CED) tone, select yes. Otherwise, select no.

The default is yes.

FAX V21 Detect Enable

To enable detection of the fax v.21 signal, select yes. Otherwise, select no.

The default is yes.

FAX Passthru Codec

Select the codec for fax passthrough, G711u or G711a.

The default is G711u.

DTMF Process INFO

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

The default is yes.

FAX Codec Symmetric

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

The default is yes.

FAX Passthru Method

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

The default is NSE.

DTMF Tx Method

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

The default is Auto.

FAX Process NSE

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

The default is yes.

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.

The default is None.

Release Unused Codec

This feature allows the release of codecs not used after codec negotiation on the first call, so that other codecs can be used for the second line. To use this feature, select yes. Otherwise, select no.

The default is yes .

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.

The default is 1 .

FAX Tone Detect Mode

If you want the Gateway to detect the fax tone whether the Gateway is a caller or callee, select caller or callee. If you want the Gateway to detect the fax tone only if the Gateway is the caller, select caller only. If you want the Gateway to detect the fax tone only if the Gateway is the callee, select callee only.

This parameter has three possible values:

caller or callee - The WRP500 will detect FAX tone whether it is callee or caller

caller only - The WRP500 will detect FAX tone only if it is the caller

callee only - The WRP500 will detect FAX tone only if it is the callee

The default is caller or callee.

FAX Enable T38

Set to yes to enable fax T.38 mode

FAX T38 ECM Enable

Set to yes to enable T38 error correction mode

Dial Plan section

The default dial plan script for each line is as follows:

(*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxx|xxxxxxxxxxxx.).

These tables describe the fields in the Dial Plan section of the Voice tab > Line page, which provide the syntax for a dial plan expression.

Dial Plan Entry

Functionality

*xx

Allow arbitrary 2 digit star code

[3469]11

Allow x11 sequences

0

Operator

00

International Operator

[2-9]xxxxxx

US local number

1xxx[2-9]xxxxxx

US 1 + 10-digit long distance number

xxxxxxxxxxxx.

Everything else (International long distance, FWD, ...)

Field

Description

Dial Plan

Dial plan script for this line.

The default is (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.)

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

Example 1:

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

Example 2:

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

Example 3:

[39]11<:@gw0>

Enable IP Dialing

Enable or disable IP dialing.

If IP dialing is enabled, one can dial [user-id@]a.b.c.d[:port], where ‘@’, ‘.’, and ‘:’ are dialed by entering *, user-id must be numeric (like a phone number) and a, b, c, d must be between 0 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.

The default is no .

Emergency Number

Comma separated list of emergency number patterns. If outbound call matches one of the pattern, the WRP500 will disable hook flash event handling. The condition is restored to normal after the phone is on-hook. Blank signifies no emergency number. Maximum number length is 63 characters.

The default is blank.

FXS Port Polarity Configuration section

This table describes the fields in the FXS Port Polarity Configuration section of the Voice tab > Line page.

Field

Description

Idle Polarity

Polarity before a call is connected: Forward or Reverse.

The default is Forward .

Caller Conn Polarity

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

The default is Forward .

Callee Conn Polarity

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

The default is Forward .

User page

You can use this page to configure the user settings. This page includes the following sections:

- Call Forward Settings section

- Selective Call Forward Settings section

- Speed Dial Settings section

- Supplementary Service Settings section

- Distinctive Ring Settings section

- Ring Settings section

When a call is made from Line 1 or Line 2, the WRP500 uses the user and line settings for that line; there is no user login support. Per user parameter tags must be appended with [1] or [2] (corresponding to line 1 or 2) in the configuration profile. It is omitted below for readability.

Call Forward Settings section

This table describes the fields in the Call Forward Settings section of the Voice tab > User page.

Field

Description

Cfwd All Dest

Forward number for Call Forward All Service

The default is blank.

Cfwd Busy Dest

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

The default is blank.

Cfwd No Ans Dest

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

The default is blank.

Cfwd No Ans Delay

Delay in sec before Call Forward No Answer triggers. Same as Cfwd All Dest.

The default is 20 .

Selective Call Forward Settings section

This table describes the fields in the Selective Call Forward Settings section of the Voice tab > User page.

Field

Description

Cfwd Sel1- 8 Caller

Caller number pattern to trigger Call Forward Selective 1, 2, 3, 4, 5, 6, 7, or 8.

The default is blank.

Cfwd Sel1 - 8 Dest

Forward number for Call Forward Selective 1, 2, 3, 4, 5, 6, 7, or 8.

Same as Cfwd All Dest.

The default is blank.

Block Last Caller

ID of caller blocked via the Block Last Caller service.

The default is blank.

Accept Last Caller

ID of caller accepted via the Accept Last Caller service.

The default is blank.

Cfwd Last Caller

The Caller number that is actively forwarded to Cfwd Last Dest by using the Call Forward Last activation code

The default is blank.

Cfwd Last Dest

Forward number for the Cfwd Last Caller parameter.

Same as Cfwd All Dest.

The default is blank.

Speed Dial Settings section

This table describes the fields in the Speed Dial Settings section of the Voice tab > User page.

Field

Description

Speed Dial 2-9

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

The default is blank.

Supplementary Service Settings section

The WRP500 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 WRP500.

This table describes the fields in the Supplementary Service Settings section of the Voice tab > User page.

Field

Description

CW Setting

Call Waiting on/off for all calls.

The default is yes .

Block CID Setting

Block Caller ID on/off for all calls.

The default is no .

Block ANC Setting

Block Anonymous Calls on or off.

The default is no .

DND Setting

DND on or off.

The default is no .

CID Setting

Caller ID Generation on or off.

The default is yes .

CWCID Setting

Call Waiting Caller ID Generation on or off.

The default is yes .

Dist Ring Setting

Distinctive Ring on or off.

The default is yes .

Secure Call Setting

If yes, all outbound calls are secure calls by default.

The default is no .

Message Waiting

This value is updated when there is voice mail notification received by the WRP500. The user can also manually modify it to clear or set the flag. Setting this value to yes can activate stutter tone and VMWI signal. This parameter is stored in long term memory and will survive after reboot or power cycle.

The default is no .

Accept Media Loopback Request

Controls how to handle incoming requests for loopback operation. Choices are: Never , Automatic , and Manual , where:

- never —never accepts loopback calls; reply 486 to the caller

- automatic —automatically accepts the call without ringing

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

The default is Automatic .

Media Loopback Mode

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

Note that if the WRP500 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 is Media .

Note that if the WRP500 answers the call, then the loopback type is determined by the caller (the WRP500 always picks the first loopback type in the offer if it contains multiple types.)

Distinctive Ring Settings section

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.

This table describes the fields in the Distinctive Ring Settings section of the Voice tab > User page.

Field

Description

Ring1 - 8 Caller

Caller number pattern to play Distinctive Ring/CWT 1, 2, 3, 4, 5, 6, 7, 8.

The default is blank .

Ring Settings section

This table describes the fields in the Ring Settings section of the Voice tab > User page.

Field

Description

Default Ring

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

The default is 1 .

Default CWT

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

The default is 1 .

Hold Reminder Ring

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

The default is 8 .

Call Back Ring

Ring pattern for call back notification.

The default is 7 .

Data Fields

This appendix describes the fields for the data parameters. After you log in, you can view or perform configuration from these tabs in the GUI:

- Quick Setup

- Interface Setup

- Network Setup

- Voice

- VPN

- Administration

- Diagnostics

- Status

Interface Setup module

The Interface Setup module includes these pages:

- Interface Setup > WAN

- Interface Setup > LAN

- Interface Setup > Wi-Fi Settings

- Interface Setup > Management Interface

Interface Setup > WAN page

Interface Setup > WAN > Internet Setup

From the Interface Setup > WAN > Internet Setup page, you can perform this configuration:

- Add a new WAN interface

- Edit an existing AN interface

- Configure a WAN interface

Add a New WAN Interface

Click the plus symbol to the right of the Ethernet WAN1 link.

Edit an Existing WAN Interface

Click the pen symbol to the right of the existing interface.

Configure WAN Interface

Either add or edit a WAN interface, The user sees a window with the fields that are described in the table that follows.

To save your settings, click the Submit button.

Field

Description

WAN

The interface ID (not applicable). This value cannot be changed.

VLAN ID

The ID for the VLAN (not applicable). VLAN 0, is used for the WAN interface, and this value cannot be changed.

Connection Type

Choose the connection type as required by your Internet Service Provider (ISP):

- Automatic Configuration - DHCP

- Static IP

- PPPoE (for ADSLuser)

- PPTP

- L2TP

Automatic Configuration - DHCP

This type of connection is often used with cable modems. Select this option if your ISP did not assign a static IP address to your account and instead uses Dynamic Host Control Protocol (DHCP) to assign an IP address dynamically. No other information is required for this selection.

Static IP

Select this option if your ISP provides you with a static IP address. Enter the following required information as provided by your ISP: Internet IP Address, Subnet Mask, and Default Gateway IP address. Optionally, you can enter the IP addresses of up to three Domain Name System (DNS) servers, or leave the fields blank to allow a DNS server to be chosen dynamically. DNS servers translate website names such as www.cisco.com into routable IP addresses.

- Internet IP Address and Subnet Mask –This is the router IP address and subnet mask as seen by external users on the Internet (including your ISP). If your Internet connection requires a static IP address, then your ISP will provide you with a Static IP Address and Subnet Mask.

- Default Gateway –Your ISP will provide you with the Gateway IP Address.

- DNS 1-3 –The Domain Name System (DNS) is the method by which the Internet translates domain or website names into Internet addresses or URLs. Your ISP will provide you with at least one DNS Server IP Address. If you wish to use another, type that IP Address in one of these fields. You can enter up to three DNS Server IP Addresses here. The router will use these for quicker access to functioning DNS servers.

PPPoE (for ADSLuser)

Select this option if your ISP uses PPPoE (commonly with DSL services). Enter the User Name and Password for your ISP account. If required by your ISP, also enter the Service Name. Finally, choose either the Keep Alive or Connect On Demand option. With Connect on Demand, the router opens a connection only when a user attempts to connect to the Internet. The connection is automatically terminated if there is a period of inactivity longer than the specified Max Idle Time (in minutes). This option is recommended if your billing is based on the time that you are connected. Alternatively, the Keep Alive option enables the router to send messages to keep the connection permanently open, regardless of the level of Internet activity by your users.

- User Name and Password –Enter the User Name and Password you use when you log on to your ISP through a PPPoE connection.

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

- Connect on Demand –You can configure the Router to terminate the Internet connection after a specified period of inactivity (Max Idle Time). If your Internet connection has been terminated due to inactivity, Connect on Demand enables the router to automatically re-establish your connection as soon as you attempt to access the Internet again. If you wish to activate Connect on Demand, click the radio button. If you want your Internet connection to remain active at all times, click the radio button next to Keep Alive. Otherwise, enter the number of minutes you want to elapse before your Internet connection terminates.

- Keep Alive –This option keeps you connected to the Internet indefinitely, even when your connection sits idle. To use this option, click the radio button next to Keep Alive. The default Redial Period is 30 seconds (that is, the router checks the Internet connection every 30 seconds).

MTU

Size, in bytes, of the largest packet that can be sent through the network. This value is typically 1500 bytes, however it might need to be lower for some broadband services. Check with your service provider for specific requirements.

Interface Setup > WAN > Internet Option

Some ISPs may require the following information. Enter this information only if your ISP instructs you to do so.

Field

Description

Host Name

A host name for the WRP500. Some service providers, usually cable service providers, require a host name and a domain name as identification. In most cases, these fields can be left blank.

Domain Name

A domain name for the WRP500. Some service providers, usually cable service providers, require a host name and a domain name as identification. In most cases, these fields can be left blank.

IPv4 Static DNS 1 - 3

Optionally, enter the IP addresses for up to three Domain Name System (DNS) servers.

Scheduled WAN Reconnect

Enabled this feature will cause all WAN connections to be restarted at the specified Reconnect Time.

Reconnect Time

Set the reconnect time by hour and minute for Scheduled WAN Reconnect feature.

Internet Setup > WAN > Mobile Network

Field

Description

Global Settings

Connect Mode

Select Auto to enable your 3G USB modem to establish a connection automatically. Select Manual to connect or disconnect your mobile connection manually. Please note that Ethernet Connection Recovery and Interface Connection Failover will work only if the Connection Mode is set to Auto. If you select Auto, you must select either Connect on Demand and Keep Alive.

- Auto/Manual

Select Auto to enable your modem to establish connection automatically. Select Manual to connect or disconnect your modem connection manually. Please note that Ethernet Connection Recovery and Interface Connection Failover will work only if the Connection Mode is set to Auto.

- Connect on Demand

Select this option to enable the router to terminate the Internet connection after it is inactive for a specified period of time (Max Idle Time). If your Internet connection is terminated due to inactivity, Connect on Demand enables the modem to automatically re-establish a terminated connection when a user attempts to access the Internet again. In the Max Idle Time field, enter the number of minutes of idle time that can elapse before your Internet connection terminates. The default Max Idle Time is 5 minutes.

- Keep Alive

Select this option to enable the router to check your Internet connection at the specified interval (Redial Period). If you are disconnected, then the router will automatically re-establish your connection. In the Redial Period field, specify how often you want the router to check the Internet connection. The default Redial Period is 30 seconds.

Tunnel Protocol

The Tunnel Protocol (PPTP/L2TP) could be supported via 3G USB modem by these simple instructions.

- None

Select this option to disable the Tunnel Protocol support. The option is used by default.

- PPTP/L2TP

Select one of the options to enable the PPTP or L2TP service you want to use. You will need to provide the server IP address, user name, and password. If you select 'None', the service would not be applied.

Card Status

This field shows the current modem connection status as Detecting, Connecting, or Connected. If your Connect Mode is Manual, there will be a button for you to click to connect or disconnect your Modem.

Mobile Network Setup

Configure Mode

Select Auto to allow the router to automatically detect which card model was inserted and which carrier is available. Select Manual to set up the connection manually. To allow the router to automatically configure modem and mobile network settings, use the default setting, Auto.

Card Model

The data card model that is inserted in the USB drive. The mobile network service provider for Internet connection. This setting is required when you are using HSDPA/UMTS/GPRS Internet service.

Access Point Name (APN)

The Internet network to which the mobile device is connecting. Enter the access point name provided by your mobile network service provider.

Dial Number

The dial number for the Internet connection. Enter the dial number provided by your mobile network service provider.

User Name/ Password

Enter the user name and password provided by your mobile network service provider.

SIM PIN

The PIN code associated with your SIM card. Enter your SIM PIN number here.

Server Name

The name of the server for the Internet connection

Authentication

The type of authentication used by your service provider. Select your authentication type. If you do not know which type to use, use the default setting, Auto.

Service Type

Select the most commonly available type of mobile data service connection based on your area service signal. If your location supports only one mobile data service, you may set up for enhanced build up connection. The first selection will always search for HSPDA/3G/UMTS service or switch to GPRS automatically only when it is available.

LTE Service

LTE (Long-Term Evolution), commonly marketed as 4G LTE, is a standard for wireless communication of high-speed data for mobile phones and data terminals.Select your LTE service. If you do not know which service to use, use the default setting, Auto

Internet Setup > WAN > Multi-WAN Config

Field

Description

Failover

Connection Failover

This feature ensures that the Internet connection is always connected through a stable WAN link. When this option is enabled, the WRP500 will first bring up the highest priority WAN interface. If the validation site associated with the WAN is unreachable, WRP500 will try to bring up the next priority WAN if available, and change system default route to that WAN. Once the validation site associated with higher priority WAN interface is reachable, WRP500 will change system default route back to the higher priority WAN interface and stop lower priority WAN connection. When this option is disabled, all WAN interfaces will try to establish the connection, and system default route will set to the highest priority WAN interface. Load balance feature is available at this time.

Failover Check Interval

Specify the time interval at which the WRP500 detects the status of the Internet connection. The default timeout interval is 60 seconds.

Failover Ping timeout

Specify the timeout value that WRP500 wait validation site response the ping request. The default timeout interval is 5 seconds.

Failover Ping Retries

Specify the retry value that validation site not respond the ping request. The default retry value is 1.

Failback after N Check Interval Successes

Specify how many successful responses from validation site the WRP500 recovery back to the high priority WAN.

Connection Validation Site

An IP address to use as a ping target to detect the status of the Internet connection. By default the WRP500 pings the gateway associated with the binding priority WAN. You may specify a different IP address as a target here.

WAN Interface

This summary provides information on the current status of the Ethernet Internet connection and the Mobile Network connection. You can click the hyperlink in the Status column to view the interface details. You may also configure the interface priority by using the Priority pull-down menu. If USB_Modem is the priority one and shows status “Connected: Validation site unreachable," configure a valid IP address in the Priority 1 WAN field.

WAN Interface Detail

List WAN information related to WAN Interfaces table. The information includes WAN interface ID, IP address, net mask and gateway address.

Load Balance

WAN Load Balancing

Enable or disable load balance. This feature is only available when Failover is disabled.

Weight

Specify the weight value associated with each WAN interface while running load balance. The valid value is between 0~99. 0 means the WAN interface will not join load balance.

Interface Setup > LAN page

Interface Setup > LAN > DHCP Server

Field

Description

DHCP Server

Add Entry

After clicking the Add Entry button, you can create another DHCP Server Pool. To edit the settings for an existing DHCP server pool, click the pencil icon.

DHCP List

Name DHCP Name, Default is DHCPRule_1(Default LAN) and DHCPRule_voice.

VLAN VLAN ID, The default is 1 and 100 .

DHCP Details

Click an entry in the DHCP List to see the details in the DHCP table

Router IP

DHCP Name

Label which identifies this DHCP Server configuration and is used to assign the service to a VLAN interface.

Local IP Address/Subnet Mask

IP address and subnet mask used to configure the VLAN interface to which this DHCP rule is applied.

DHCP Server Setting

DHCP Mode

To select this DHCP pool run as DHCP Server or DHCP Relay agent. Please note, DHCP Relay only works when the NAT function is disabled.

DHCP Server

Show DHCP Reservation

Click this button to review and modify the DHCP reservations. Click the button again to hide the reservation tables.

Select Clients from DHCP Tables

Shows the clients that are currently receiving IP addresses from the DHCP server. If you want to reserve the currently assigned IP address for exclusive use by a client, check the Select box and click Add. The client appears in the Clients Already Reserved table.

Manually Adding Client

To reserve an IP address for a client, enter a client name and an IP address that you want to reserve for the client. Then enter the MAC address of the client and click Add. The client appears in the Clients Already Reserved table.

WAN Interface

Choose the WAN Interface from which the related DHCP information, specifically DNS, is obtained.

Default Gateway

Enter the IP address of the default gateway to be used by clients of this pool. If the field is 0.0.0.0. the VLAN Local IP Address is used as the default gateway.

Option 66

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

- Local TFTP Server: The WRP500 uses its own TFTP server to source provisioning files so it returns its own local IP address to the client.

- Remote TFTP Server: If the WRP500 was configured by using this method, it uses the server information it received through option 66 on its WAN interface in response to local client requests.

- 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 WRP500 will also accept and offer a full URL including protocol, path and filename to meet to requirements of specific clients.

Option 67

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

Option 159

Provides a configuration URL to clients requesting 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

Option 160

Provides a configuration URL to clients requesting 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

DNS Proxy

If DNS proxy is enabled, local clients are offered the WRP500 Local IP Address to use for DNS requests. The WRP500 then proxies these requests to the DNS servers it was configured with.

If DNS proxy is disabled, then DHCP clients will be offered DNS server information based on the following:

If the Static DNS field is configured, then that server alone will be offered to clients.

If the Static DNS field is not configured up to three servers are offered, first from the global Internet Options static configuration and then from the WAN interface nominated above.

Starting IP Address

Enter an IP address of the first address in this pool.

Maximum DHCP Users

Enter the maximum number of devices that you want the DHCP server to assign IP addresses to.This number cannot be greater than 256.

Client Lease Time

Amount of time an address is leased to a client. Enter the amount of time, in minutes, for the lease. The default is 0 minutes, which means one day. Enter 9999 to assign an infinite lease.

WINS

The Window Internet Naming Service (WINS) manages the window's host name to address resolution. If you use a WINS server, enter the IP address of the server here. Otherwise, leave this field blank.

DHCP Relay

Remote DHCP Server

Set the DHCP server IP address that DHCP message will be relayed to.

Interface Setup > LAN > VLAN Settings

Field

Description

VLAN Settings

Add Entry

Click the Add Entry button to create another VLAN.

VLAN List

- Name—VLAN Name. The default is data_Lan and voice_Lan.

- ID—VLAN ID, The default is data_Lan : 1 and voice_Lan : 100.

- Address Type—LAN Address Type. The default is data_Lan and voice_Lan : DHCP Server Pool.

- Voice—Voice, The default is data_Lan: disabled and voice_Lan : enabled.

- Membership—Membership, The default is data_Lan: LAN Port 1-4 and SSID1, voice_Lan : LAN Port 1-4 and SSID2.

VLAN Details

Select one item the VLAN List, the Detail of VLAN table will show all VLAN information.

VLAN – Add

VLAN Name

Enter your VLAN Name.

VLAN ID

Enter an identification number for the VLAN.

Note that VLAN ID 0~2, and 4080~4095 are reserved for internal interfaces, and cannot be set as the manual VLAN ID.

Voice VLAN

Click this box if you want voice applications to use this VLAN.

Note All traffic from a voice VLAN follows the voice default route specified in WAN interface configuration unless there is policy based routing configured for the voice VLAN. Policy based routing takes precedence over the default route. There are no implicit QoS settings for voice VLAN. You will need to change these accordingly.

Role

When bridging LAN ports with a WAN interface, the VLAN role will control how the associated IP interface is created.

- Select the WAN role to create the IP interface as a subinterface of the selected Ethernet WAN. The resulting VLAN will be a layer 2 broadcast domain on the outside of the firewall.

- Select the LAN role to create the IP interface, if required, as a LAN VLAN. VLANs created without WAN interfaces are automatically created with the LAN role.

IPv4 Address Type

Address type determines the way in which the VLAN IP interface is configured.

- Choose None if an IP interface is not required. This would typically be the case when bridging ports only.

- Choose Static IP Address to manually define an address for the interface.

- Choose Dynamic IP Address to request an address from a DHCP server on the local network.

- Choose DHCP server to enable a previously configured DHCP Server service on this interface. In this case, the VLAN IP address will be derived from the DHCP Server configuration.

Available Interface

The interfaces that are available to be added to the VLAN. To move an interface to the Added Interface list, click the interface, and then click the right-arrow button (>). To move all of the interfaces at once, click the double right arrow button (>>).

Added Interface

The interfaces that were selected as members of the VLAN bridge. If you want to remove an interface from this list, click the interface and then click the left arrow button (<). To remove all of the interfaces at once, click the double left-arrow button (<<).

Interface Setup > LAN > Port Settings

Field

Description

Port Settings

Port List

- Interface

Show Port Interface.

- Mode

Describes the currently configured behavior of the port.

- Desktop mode: Provides attached devices with access to a single data VLAN for which the WRP500 provides DHCP services. Incoming traffic from the host can be tagged or untagged. Outgoing traffic to the host will be untagged.

- IP Phone + Desktop mode: The port is configured with a data VLAN for native access and a voice VLAN for use with an attached IP Phone. CDP is used to communicate voice VLAN information to the phone.

- Switch/AP mode: The port is configured to be part of multiple VLANs (any combination other than 1 data and 1 voice VLAN) for the purposes of trunking to either a switch or wireless access point.

- Generic: The port is configured for layer 2 bridging mode only.

- Enabled Flow Control

Mechanism for temporarily stopping the transmission of data on this physical interface. 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 send a PAUSE frame, which halts the transmission of the sender for a specified period of time. To enable this feature, check the box. The default setting is Enabled.

- Speed Duplex (Ethernet Port 1~4)

Choose the duplex mode. You can select from Autonegotiate, 10 Half, 10 Full, 100 Half,100 Full,1000 Half and 1000 Full. The default is Auto-negotiate.

Port

Defines the quality of service trust settings for the port. The default setting is untrusted.

- If the port is not trusted, the queuing priority for incoming traffic is defined by the port priority setting.

- If the port is trusted, the queuing priority for the traffic is determined by 802.1p priority (CoS to Queue) if present, or IP priority (DSCP to Queue) if not. If neither priority is available, the queuing priority is set based on the port setting.

Port/Access VLAN

Select the native VLAN (PVID) for this port. The dropdown list includes all VLAN IDs that were configured on the VLAN Settings page.

Voice VLAN

When the VLAN mode is IP Phone + Desktop, the voice VLAN ID is shown. This value is informational only.

Priority

Set a priority for unmarked, or untrusted traffic received on this port. By default, the priority is set to 0. A higher number indicates a higher priority.

Interface Setup > LAN > STP

Field

Description

STP

Bridge Priority

Bridge priority is used to influence what bridge becomes the STP root. The bridge with the lowest value in the network will be elected as the root. Valid Bridge Priorities range from 0 through 61440, in steps of 4096. The default value is 32768.

Forward Delay

Note Forward Delay, Hello Time, and Max Age are configuration settings sent by the root bridge to all other bridges to define the current STP configuration. If the WRP500 is not elected as the root, the active timer values might be different to those configured here.
Forward Delay is the time spent in the listening and learning state. This time is equal to 15 seconds by default, but you can adjust the time to be between 4 and 30 seconds. Base IEEE 802.1D Standard, to support interoperability with legacy Bridges, a Bridge shall enforce the following relationship:

2 x (Forward_Delay - 1.0 seconds) >= Max_Age

Max_Age >= 2 x (Hello_Time + 1.0 seconds)

Hello Time

The Hello Time is the time between each Bridge Protocol Data Unit (BPDU) that is sent by a bridge. This time is equal to 2 seconds by default, but you can adjust the time to be between 1 and 10 seconds. Base IEEE 802.1D Standard, to support interoperability with legacy Bridges, a Bridge shall enforce the following relationship:

2 x (Forward_Delay - 1.0 seconds) >= Max_Age

Max_Age >= 2 x (Hello_Time + 1.0 seconds)

Max Age

The Max Age timer defines how long bridges will wait after receiving the last hello message before assuming that the layer 2 topology has changed. At this point the current spanning tree configuration is discarded and the new topology is discovered. This time is 20 seconds by default, but you can adjust the time to be between 6 and 40 seconds. Base IEEE 802.1D Standard, to support interoperability with legacy Bridges, a Bridge shall enforce the following relationship:

2 x (Forward_Delay - 1.0 seconds) >= Max_Age

Max_Age >= 2 x (Hello_Time + 1.0 seconds)

Interface Setup > Wi-Fi Settings

Interface Setup > Wi-Fi Settings > Basic Wireless Settings

Field

Description

Wireless Network

Operating Radio

Select Radio 1 (2.4 GHz) or Radio 2 (5 GHz) to specify which radio to configure. The rest of the settings on this tab apply to the radio you select in this field. Be sure to configure settings for both radios.

Wireless Table

Wireless Network Name (SSID)

The default wireless network uses this name: “cisco-radio1-data” To rename the default wireless network, enter a unique Wireless Network Name, which is case-sensitive and must not exceed 32 characters (use any of the characters on the keyboard).

To create a second wireless network, enter a unique Wireless Network Name in the SSID2 setting. (To activate this network, select Network Enabled.)

Note: Your ISP or ITSP may control the SSID2 settings. Contact your ISP or ITSP for more information.

Broadcast Network Name

When wireless clients survey the local area for wireless networks to associate with, they detect the SSID broadcast by the Router. If you want to broadcast the SSID, leave the check box selected. If you do not want to broadcast the SSID, deselect the check box.

Enabled Network

To enable the wireless network, select the check box. To disable the wireless network, deselect the check box.

WPS Hardware Button

Security

These settings configure the security of your wireless network.

Click “Edit” to configure SSID security

Wireless Security

Security Mode

Select the security method for your wireless network. Proceed to the appropriate instructions. If you do not want to use wireless security, use the default, Disabled.

- WEP

- WPA2 Personal

- WPA/WPA2-Mixed Personal

- WPA2 Enterprise

- WPA/WPA2 Enterprise

Interface Setup > Wi-Fi Settings > Wi-Fi Protected Setup

Field

Description

Select a SSID

From this drop-down menu, you can decide the WPS settings apply to which SSID. The default is SSID1.

Wi-Fi Protected SetupTM

Select disabled if you don't want to use the Wi-Fi Protected Setup. The default is Disabled. There are three methods available. Use the method that applies to the client device; you are configuring.

- Method 1 Use this method if your client device has a Wi-Fi Protected Setup button.

a. Click or press the Wi-Fi Protected Setup button on the client device.

b. Click the Wi-Fi Protected Setup button on this screen.

c. After the client device has been configured, click

d. OK.

Then refer back to your client device or its documentation for further instructions.

- Method 2 Use this method if your client device has a Wi-Fi Protected Setup PIN number.

a. Enter the PIN number in the field on this screen.

b. Click

c. Register.

d. After the client device has been configured, click

e. OK. Then refer back to your client device or its documentation for further instructions.

- Method 3 Use this method if your client device asks for the Router's PIN number.

a. Enter the PIN number listed on this screen. (IT is also listed on the label on the bottom of the Router.)

b. After the client device has been configured, click

c. OK. Then refer back to your client device or its documentation for further instructions.

The Wi-Fi Protected Setup Status, Network Name (SSID), Security are displayed at the bottom of the screen.

Interface Setup > Wi-Fi Settings > Advanced Wireless Settings

Field

Description

Operating Radio

Radio 1 and Radio 2 can be selected, after configure radio 1, you can select radio 2 to continue configuration.

RTS Threshold

The router sends Request to Send (RTS) frames to a receiving station and negotiates the sending of a data frame. After receiving an RTS, the wireless station responds with a Clear to Send (CTS) frame to acknowledge the right to begin transmission. If you encounter inconsistent data flow, you can adjust this threshold. Only minor reduction of the default value, 2347, is recommended. If a network packet is smaller than the preset RTS threshold size, the RTS/CTS mechanism will not be enabled. The RTS Threshold value should remain at its default value of 2347.

AP Isolation

This feature isolates all wireless clients and wireless devices from one another. Wireless devices will be able to communicate with the router but not with other wireless devices on the network. To use this function, select Enabled. AP Isolation is disabled by default.

Basic Rate

The Basic Rate setting is not actually one rate of transmission but a series of rates at which the Router can transmit. The Router will advertise its Basic Rate to the other wireless devices in your network, so they know which rates will be used. The Router will also advertise that it will automatically select the best rate for transmission. The default setting is Default, which allows the Router to transmit at all standard wireless rates (1-2Mbps, 5.5Mbps, 11Mbps, 18Mbps, and 24Mbps). Other options are 1-2Mbps, for use with older wireless technology, and All, which allows the Router to transmit at all wireless rates.

DTIM Interval

This value, between 1 and 255, indicates the interval of the Delivery Traffic Indication Message (DTIM). A DTIM field is a countdown field informing clients of the next window for listening to broadcast and multicast messages. When the Router has buffered broadcast or multicast messages for associated clients, it sends the next DTIM with a DTIM Interval value. Its clients hear the beacons and awaken to receive the broadcast and multicast messages. The default value is 1

Fragmentation Threshold

This value specifies the maximum size for a packet before data is fragmented into multiple packets. If you experience a high packet error rate, you may slightly increase the Fragmentation Threshold. Setting the Fragmentation Threshold too low may result in poor network performance. Only minor reduction of the default value is recommended. In most case, it should remain at its default value of 2346.

Beacon Interval

Enter a value between 20 and 1,024 milliseconds. The Beacon Interval value indicates the frequency interval of the beacon. A beacon is a packet broadcast by the router to synchronize the wireless network. The default value is 100.

Power Control

Form this drop-down menu, you can choose high, middle, or low value to specify the range of the wireless network. This option will determine the available distance. The default is high which is a normal power level.

PMF Capable

The 802.11w protocol applies only to a set of robust management frames that are protected by the Management Frame Protection (PMF) service. These include Disassociation, Deauthentication, and Robust Action frames. When PMF capable enabled, it is to be used if the client supports 802.11w.

PMF Required

Enabled PMF required will ensure that the clients that do not support 802.11w cannot associate with the WLAN.

PMF SHA256

Enable or Disable SHA-256 key derivation functionality.

Multicast Power Save

Enable or Disable Multicast Power Save.

Interface Setup > Wi-Fi Settings > WMM Setting

Field

Description

Wireless

Operating Radio

Radio 1 and Radio 2 can be selected, after configure radio 1, you can select radio 2 to continue configuration.

WMM Support

WMM provides Quality of Service features to support voice and video applications. To enable WMM, keep the default setting, Enabled. Otherwise, choose Disabled.

No Acknowledgement

When this option is enabled, the router does not resend data if an error occurs. To enable this feature, keep the default setting, Disabled. Otherwise, choose Enabled.

Interface Setup > Management Interface

Field

Description

List of Management Interface

IP Address

Enter the IP Address to use for the loopback test. If IP Address and WAN IP Address or LAN IP Address are the same, it is unavailable.

Network Setup module

The Network Setup module includes these pages:

- Network Setup > Routing

- Network Setup > NAT

- Network Setup > QoS

- Network Setup > Firewall

- Network Setup > PPPoE Relay

- Network Setup > DDNS

- Network Setup > DMZ

- Network Setup > IGMP

- Network Setup > UPnP

- Network Setup > CDP

- Network Setup > LLDP

- Network Setup > DNS Spoofing

Network Setup > Routing page

Network Setup > Routing > Static Routes > IPv4

Field

Description

Add Entry

After clicking the Add Entry button, it will create another Static Route.

Static Routing list

- Name

Show all routes of name.

- Interface

Show all routes of interface.

Static Routing Details

Select one entry of Static Routing list, Defaults of Static Routing Details will show all Information. (Link Route Name, Destination IP Address, Subnet Mask, Gateway, Interface).

Enter Route Name

Enter a net Static Routing Name.

Destination

The Destination IP Address is the address of the network or host to which you want to assign a static route.

Subnet Mask

The Subnet Mask determines which portion of an IP address is the network portion, and which portion is the host portion.

Gateway

The IP address of the gateway device that allows for contact between the Router and the network or host.

Routing Table

- Destination LAN IP

The Destination IP Address is the address of the network or host to which the static route is assigned.

- Subnet Mask

The Subnet Mask determines which portion of an IP address is the network portion, and which portion is the host portion.

- Gateway

This is the IP address of the gateway device that allows for contact between the Router and the network or host.

- Interface

This interface tells you whether the Destination IP Address is on the LAN (internal wired and wireless networks), the Internet (WAN).

Network Setup > Routing > RIP > IPv4

Field

Description

RIP

Routing Information Protocol is used for dynamic routing. You can enable this protocol to allow the specified interfaces to automatically adjust to physical changes in the network's layout and to exchange routing tables with other router. The router determines the network packets' route based on the fewest number of hops between the source and destination. To enable the Dynamic Routing feature, select Enabled then enter the RIP settings, and enable RIP on the interfaces where you want to use this feature. To disable the Dynamic Routing feature for all data transmissions, use the default setting, Disabled.

RIP Version

To limit the types of packets that can be transmitted, choose Version 1 or Version 2. Alternatively, choose RIP v1/v2 to allow both Version 1 and Version 2 packets to be transmitted.

RIP Timer

RIP uses timers to regulate its performance. These include a routing-update timer, a route-timeout timer, and a route-flush timer.

- Update

Specify the rate at which the router sends routing updates. The default is 30 seconds

- Timeout

Specify the rate at which the router expects to receive routing updates from each router in the routing table. If this value is exceeded, the route is declared unreachable. The route is not removed from the routing table until the route flush timer expires.

- Flush

Specify the maximum period that the router will wait for an update before removing a route from the routing table.

RIP List

This list displays the RIP settings for the WAN interface (WAN1) and each VLAN. To edit the settings, click the pencil icon.

- Interface

Show RIP default interface.

- RIP Enable

- Passive

- Authentication

RIP Network

- Network Address

Specifies the IP Address and Subnet mask for the entry.

Network Setup > Routing > Intervlan Routing

Field

Description

Intervlan Routing

Configuring VLANs helps control the size of the broadcast domain and keeps local traffic local. However, when an end station in one VLAN needs to communicate with an end station in another VLAN, Intervlan communication is required. This communication is supported by Intervlan routing. To enable this feature, keep the default setting, Enabled . To disable this feature, choose Disabled .

Network Setup > Routing > Policy Routing

Field

Description

Add Entry

Name

Specify the name for this policy route rule

Incoming Interface

Select LAN interface to apply for a rule. Any means the OUT Interface for this rule applied for all LAN interfaces

Source IP Address

Matches the source IP address from which packets are addressed to this rule.

Subnet Mask

Defines the source IP address wildcard mask. Masks specify which bits are used and which bits are ignored. A mask of 255.255.255.255 indicates that all the bits are important. A mask of 0.0.0.0 indicates that all the bits will be ignored. Therefore, if an Source IP Address is specified but source subnet mask specified to 0.0.0.0, the rule will regards it as 0.0.0.0/0 (all) address.

Destination IP Address

Matches the destination IP address to which packets are addressed to this rule.

Subnet Mask

Defines the destination IP address wildcard mask. Masks specify which bits are used and which bits are ignored. A mask of 255.255.255.255 indicates that all the bits are important. A mask of 0.0.0.0 indicates that all the bits will be ignored. Therefore, if a destination IP address is specified but destination subnet mask specified to 0.0.0.0, the rule will regards it as 0.0.0.0/0 (all) address.

Port

Defines the TCP/UDP destination port to match. "Any" means port field will not be inspected. "Single" means a port is specified. "Range" means that port range are specified

Protocol

Specify the interested protocol for this rule. Default is "Any", which, means protocol filed filter will be disabled and all kind of protocol are going to inspect. Beside, user can specify UDP, TCP

DSCP

Specify the DSCP number to match for this rule.

Route

Two possible output categories can be selected. One is existed VPN tunnel, and the other is existed WAN interface. When select WAN interface as output interface, an additional option can be checked: "Disable this rule if the interface is down". When this option is checked, the policy route rule will take no effect while the output interface is down (got no IP). The traffic will then fall through to match other policy route rules, or obey system's route (typically system default route).

Network Setup > NAT

Network Setup > NAT > NAT Setting

Field

Description

Address Translation

NAT

Choose the correct working mode. Use the default setting, Enabled, if the Router is hosting your network's connection to the Internet (Enabled mode is recommended for most users). Select Disabled if the Router exists on a network with other routers

Application Layer Gateways

SIP

SIP ALG can help to establish, modify, and terminate multimedia sessions (conferences) such as Internet telephony calls. The default is Enabled.

NetMeeting

NetMeeting ALG can modify RAS, Fast Start, H.245 Tunneling, Call Forwarding, RTP/RTCP and T.120 based audio, video, fax, chat, whiteboard, file transfer. Besides, it only support connected way from LAN hosts to WAN hosts at present. If you want to connected way from WAN to LAN, you need to set DMZ. The default is Enabled.

RTSP

RTSP ALG allows UDP transports to be setup properly, including RTP and RDT. The default is Enabled.

IRC

IRC ALG can allow users to send files to each other and user need connect to IRC server. The default is Enabled.

Network Setup > NAT > NAT Bypass

Field

Description

NAT Bypass

NAT Bypass Policy Setting which addressed a flexible and configurable rule matching criteria to set the matched traffic to perform pure routing while global NAT option is enabled.

Add Policy

Click the Add Policy button to create a NAT bypass rule

Policy List

- Policy Name

User specified NAT bypass rule name

- Inside Interface

This field presents user specified inside interface, which will be VLAN interface, Host IP address or Indirect Network

- Outside Interface

This field presents user specified outside interface

- Status

This field presents this NAT bypass rule is enabled or disabled

Policy Details

All detailed information will be shown by selecting one entry from the list of NAT bypass rule.

Policy Name

Specify the rule name for this rule.

Enable

Enable/disable this rule.

Inside interface

Specify the traffic source rule

- VLAN interface

Specify the VLAN that to become the NAT bypass VLAN domain. The pull down menu contain all LAN (VLAN) collection. This is the either one option between Host and Indirect Network Domain options.

- Host IP Address

Specify a host IP address that to become a routed host. This is the either one option between VLAN and Indirect Network Domain options

- Indirect Network

Specify an indirect network domain (non-VLAN) to be a routed domain. This is the either one option between VLAN and Host options.

- IP Address

Specify source IP address associated with the indirect network domain when indirect network domain option selected.

- Subnet Mask

Specify the subnet mask associated with the source IP address when indirect network domain option is selected.

Outside Interface

Specify the traffic destination rule.

- WAN interface

Select the out interface. The pull down menu contains all WAN collection.

- IP Address

Specify destination IP address

- Subnet Mask

Specify the subnet mask associated with the destination IP address.

Network Setup > NAT > Port Forwarding

Port Forwarding

Use the Port Forwarding page if your network hosts network services (Internet applications) such as World Wide Web, email, FTP, videoconferencing or gaming. For each service, you need to configure the settings to forward Internet traffic to the servers that host these services. After clicking the Add Entry button, you can create another entry for another network service. To edit an entry, click the pencil icon. Before you perform this procedure, you should reserve a DHCP addresses for each server that hosts an Internet application. Use the Interface Setup > LAN > DHCP Server page. Click Add Entry, and then click Show DHCP Reservation. You can add the server from the Select Clients table, or manually enter the client information.

Add Entry

Click the Add Entry button to create another Single Port Forwarding or Port Range Forwarding

List of Port Forwarding

- Number

- Type

Show Port Forwarding entry type is Single Port Forwarding or Port Range Forwarding.

- Status

Show Enable or Disable the entry.

- Application

Show Entry Name.

Port Forwarding Details

Select one entry from the List of Port Forwarding Details of Port Forwarding will show all Information. (like Wan Interface Name, External Port, Internal Port, Protocol, IP Address).

Port Forwarding Type

Choose Single Port Forwarding to forward traffic to a single port on the specified server, or choose Port Range Forwarding to forward traffic to a range of ports.

Single Port Forwarding

- Application Name: Choose a standard application from the drop-down list. To enter an application that is not on the list, choose Add a new name, and then enter the name of a new application.

- Enter a Name: Enter the name of the Internet application.

- WAN Interface Name: Choose the WAN interface through which the traffic is transmitted

- External Port: For single port forwarding, enter the external port number that is used by the server or Internet application. Check the Internet application's documentation for more information

- Internal Port: For single port forwarding, enter the internal port number used by the server or Internet application. Check the Internet application's documentation for more information.

- Protocol: Select the protocol TCP or UDP.

- IP Address: Enter the IP address of the server that hosts this Internet application. The server must have a static IP address, which you can set on the Interface Setup > LAN > DHCP Server page.

- Enabled: Check the box to enable the application you have defined. The default setting is unchecked (Disabled)

Port Range Forwarding

- Enter a Name: Enter the name of the Internet application

- WAN Interface Name: Choose the WAN interface through which the traffic is transmitted

- Start ~ End Port: For port range forwarding, specify the range of ports used by the server or Internet application. Enter the first port in the first box, and enter the final port in the second box to specify the range. Check the Internet application's documentation for more information.

- Protocol: Select the protocol TCP or UDP.

- IP Address: Enter the IP address of the server that hosts this Internet application. The server must have a static IP address, which you can set on the Interface Setup > LAN > DHCP Server page.

- Enabled: Check the box to enable the application you have defined. The default setting is unchecked (Disabled).

Network Setup > NAT > Port Range Triggering

Field

Description

Port Range Triggering

Use the Port Range Triggering page to allow the router to dynamically open ports for network services (Internet applications) that are hosted by individual computers. When this feature is enabled, an outbound connection from specified ports triggers the router to open other specified ports for incoming traffic.

Port Range Triggering does not require you to reserve an IP address (static IP address) for the computer that hosts the specified application. However, Port Range Triggering allows only one computer to host a service on the specified ports at one time.

Add Entry/Edit

After clicking Add Entry button, it can create another Port Range Triggering.

Port Range Triggering List

- Status

Show Enable or Disable the entry.

- Application

Show Entry Name.

Port Range Triggering Details

Select one entry of Port Range Triggering List, Details of Port Triggering will show all Information, such as WAN Interface, LAN Interface, Triggered Range, Forwarded Range, Protocol.

Application Name

Enter a name to identify the application in the Port Range Triggering List

WAN

Choose the WAN Interface for the Internet traffic

LAN

Choose the LAN where the host computer is located

Triggered Range

Enter the starting and ending port numbers of the triggered port range. When a computer makes an outbound connection from these ports, the router will open the ports that are specified in the Forwarded Range fields. Check with the Internet application's documentation for the port number(s) needed.

Forwarded Range

Enter the starting and ending port numbers of the forwarded port range. These ports will be opened when an outbound connection is made from the ports that are specified in the Triggered Range fields. Check with the Internet application documentation for the port number(s) needed.

Protocol

Select the protocol TCP or UDP.

Enabled

Click the Enabled check box to enable the applications you have defined. This is disabled (unchecked) by default.

Network Setup > QoS

Network Setup > QoS > QoS Bandwidth Control

Field

Description

QoS Bandwidth Control

Set the bandwidth priority rule for a variety of interface.

Name

Show the interface name

Enabled

To use the QoS policies you have set, select the check box. Otherwise, deselect the check box.

Upstream Bandwidth

Show the maximum bandwidth for upstream data transmissions

Strict High/ High / Medium / Normal / Low

Show the bandwidth guarantees for each priority queues

Upstream Bandwidth

Set the maximum bandwidth for upstream data transmissions.

Priority

Set the bandwidth guarantees for the priority queues.

- Strict High

Enter the guaranteed bandwidth for the Strict High Priority queue.

- High, Medium, Normal, Low

Increase the rate and bandwidth for each queue by clicking the plus (+) button, or reduce the rate and bandwidth by clicking the minus (-) button.

Network Setup > QoS > QoS Policy

Field

Description

QoS Policy

Configures the Quality of Service (QoS) settings for specified applications, devices, ports, or VLANs.

Quality of Service (QoS) ensures better service to high-priority types of network traffic, which may involve demanding, real-time applications, such as video conferencing. When you set priority, do not set all applications to High, because this will defeat the purpose of allocating the available bandwidth. If you want to select below normal bandwidth, select Low. Depending on the application, a few attempts may be needed to set the appropriate bandwidth priority.

Add Entry

Click the Add Entry button to create another QoS Policy.New create rule will have high priority than old one

List of QoS Policy

- Priority

Show the priority of entry.

- Name

Show the name of entry

QoS Details

Select one entry of List of QoS Policy, Details of QoS will show all information about QoS

Category

There are five categories available. Select one of the following: Application, MAC Address, Ethernet Port, VLAN and IP Address , then complete the fields that appear, based on your selection.

Application

- Applications: Choose a standard application from the drop-down list. To enter an application that is not on the list, choose Add a New Application, and then enter the name.

- Name: For most categories and applications, this field displays the name of the selected category or application. If you chose Add a New Application, enter the name of the application.

- LAN: Choose the VLAN that is used for this traffic

- Port Range:

– Port Range: Enter the number or range of port(s) used by the server or Internet application. Check the Internet application's documentation for more information. Also select the protocol TCP or UDP, or select Both.

– Protocol: Select the protocol TCP or UDP, or select Both

- Priority: Choose the bandwidth priority for this traffic: Strict, High, Medium, Normal, or Low, Medium is recommended

- Marking: Marking modifies the DiffServ or CoS field of the packet according to QoS Policy Rule (by Application port range, Mac, Ethernet port, VLAN and IP Address).

Note: CoS value only valid when output interface is subwan (with 802.1Q tagging).

- CoS and DiffServ: Setting Network Setup/QoS/Qos policy, this will classify the LAN to WAN packet. If traffic doesn't match these policy rules, it will use default priority setting by each Ethernet port if it is untrust mode, or classify by packet COS or DiffServ priority setting if it is trust mode.

MAC Address

- Name: Enter a name to describe this rule.

- LAN: Choose the VLAN that is used for this traffic

- MAC Address: Enter the MAC address of the device in the following format: xx:xx:xx:xx:xx:xx

- Priority: Choose the bandwidth priority for this traffic: Strict, High, Medium, Normal, or Low, Medium is recommended.

- Marking: Marking modifies the DiffServ or CoS field of the packet according to QoS Policy Rule (by Application port range, Mac, Ethernet port, VLAN and IP Address).

Note CoS value only valid when output interface is subwan (with 802.1Q tagging).

- CoS and DiffServ: Setting Network Setup/QoS/Qos policy, this will classify the LAN to WAN packet. If traffic doesn't match these policy rules, it will use default priority setting by each Ethernet port if it is untrust mode, or classify

Ethernet Port

- Name: Enter a name to describe this rule.

- LAN: Choose the VLAN that is used for this traffic

- Ethernet Choose the Ethernet port.

- Priority: Choose the bandwidth priority for this traffic: Strict, High, Medium, Normal, or Low, Medium is recommended.

- Marking: Marking modifies the DiffServ or CoS field of the packet according to QoS Policy Rule (by Application port range, Mac, Ethernet port, VLAN and IP Address).

Note CoS value only valid when output interface is subwan (with 802.1Q tagging).

- CoS and DiffServ: Setting Network Setup/QoS/Qos policy, this will classify the LAN to WAN packet. If traffic doesn't match these policy rules, it will use default priority setting by each Ethernet port if it is untrust mode, or classify by packet's COS or DiffServ priority setting if it is trust mode.

VLAN

- Name: Enter a name to describe this rule.

- LAN: Choose the VLAN that is used for this traffic

- Priority: Choose the bandwidth priority for this traffic: Strict, High, Medium, Normal, or Low, Medium is recommended.

- Marking: Marking modifies the DiffServ or CoS field of the packet according to QoS Policy Rule (by Application port range, Mac, Ethernet port, VLAN and IP Address).

Note CoS value only valid when output interface is subwan (with 802.1Q tagging).

- CoS and DiffServ: Setting Network Setup/QoS/Qos policy, this will classify the LAN to WAN packet. If traffic doesn't match these policy rules, it will use default priority setting by each Ethernet port if it is untrust mode, or classify by packet's COS or DiffServ priority setting if it is trust mode.

IP Address

- Name: Enter a name to describe this rule.

- Destination IP Address: Set the destination IP address of traffic flow that would apply QoS.

- Destination Mask: Set the subnet mask to decide the destination IP address range.

- Priority: Choose the bandwidth priority for this traffic: Strict, High, Medium, Normal, or Low, Medium is recommended.

- Marking: Marking modifies the DiffServ or CoS field of the packet according to QoS Policy Rule (by Application port range, Mac, Ethernet port, VLAN and IP Address).

Note CoS value only valid when output interface is subwan (with 802.1Q tagging).

- CoS and DiffServ: Setting Network Setup/QoS/Qos policy, this will classify the LAN to WAN packet. If traffic doesn't match these policy rules, it will use default priority setting by each Ethernet port if it is untrust mode, or classify by packet's COS or DiffServ priority setting if it is trust mode.

Network Setup > QoS > CoS To Queue

Field

Description

VLAN CoS

Specifies the VLAN (CoS) priority tag values, where zero is the lowest and 7 is the highest.

Priority

Defines the traffic forwarding queue to which the CoS priority is mapped. Where five kinds of traffic priority queues are supported.

Network Setup > QoS > DSCP To Queue

Field

Description

DiffServ

Indicates the Differentiated Services Code Point (DSCP) value in the incoming packet

Priority

Defines the traffic forwarding queue to which the DSCP priority is mapped

Network Setup > Firewall

Network Setup > Firewall > Firewall Filter

Field

Description

SPI Firewall Protection

A firewall enhances network security and uses Stateful Packet Inspection (SPI) for more review of data packets entering your network. Select Enabled to use a firewall, or Disabled to disable it.

Filter Anonymous Internet Requests

When enabled, this feature keeps your network from being "pinged," or detected, by other Internet users. It also hides your network ports. Both to make it more difficult for outside users to enter your network. This filter is enabled by default. Select Disabled to allow anonymous Internet requests.

Filter Internet NAT Redirection

This feature uses port forwarding to block access to local servers from local networked computers. This filter is disabled by default. Select Enabled to filter Internet NAT redirection, or Disabled to disable this feature.

Filter IDENT (Port 113)

This feature keeps port 113 from being scanned by devices outside of your local network. This filter is enabled by default. Select Enabled to filter port 113, or Disabled to disable this feature.

Filter DoS Attack

When enabled, this feature wards off ICMP Ping flood (ICMP echo request) and TCP SYN flood (tcp_syn cookies) attacks. The default is disabled. Check the box to enable it. The maximum rate limit for both types of flood attacks is 50 packets per second

Note: If an IP packet is destined to an IP broadcast or IP multicast destination address, your network can be used to execute a flooding DoS attack to other hosts.

Proxy

Use of WAN proxy servers may compromise your network security. Denying Filter Proxy will disable access to any WAN proxy servers. To enable proxy filtering, check the box.

Java

Java is a programming language for websites. If you deny Java, you run the risk of not having access to Internet sites created using this programming language. To enable Java filtering, check the box.

ActiveX

ActiveX is a programming language for websites. If you deny ActiveX, you run the risk of not having access to Internet sites that use this programming language. To enable ActiveX filtering, check the box.

Cookies

A cookie is data stored on your computer and used by Internet sites when you interact with them. To prevent the storage of cookies, check the box.

Filter Port

Set the Web service port for filtering Proxy/Java/ActiveX/Cookies, port 80 is used by default

Network Setup > Firewall > Firewall Filter

Field

Description

Internet Filter

- Filter Anonymous Internet Requests

- Filter Internet NAT Redirection

- Filter IDENT (Port 113

- Filter DoS Attack

Web Filter

- Proxy

- Java

- ActiveX

- Cookies

Network Setup > Firewall > IPV4 > Internet Access Control

Field

Description

Internet Access Control

Configures rules controlling users' access to the Internet

Add Entry

Click the Add Entry button to create another Internet Access Control

Internet Access Policy list

- PolicyName

Show the entry of name.

- Status

Show the entry of status.

- From LAN Interface

Show the entry of LAN interface.

- To WAN Interface

Show the entry of WAN interface.

Policy Details

Select an entry from the Internet Access Policy list, Details of Policy will show all information about Internet Access Policy.

Policy Name

Add Policy Name

Status

To enable this policy, click Enabled. To disable this policy, click Disabled. The default is Disabled

From LAN, To WAN

You can apply the rule to all traffic by choosing From All, To All, or you can limit the rule to apply only to particular interfaces, such as From VLAN1 to Ether_WAN1

Applied PCs

If you want the policy to apply only to specified PCs, click the Show Edit List button. Then you can specify individual PCs by entering the MAC address or the IP address. You can specify groups of PCs by entering up to two ranges of IP addresses

Days/Times

Choose the days and times when you want this policy to be enforced. Select the individual days, or select Everyday. Enter a range of hours by specifying the start time (From) and the end time (To), or select 24 Hours.

Blocking Everything

Check this box to block all Internet traffic that meets the criteria that you specified on this page. Uncheck this box to choose one or more of the other filtering options.

Blocking by URL and Keyword

Check this box to prevent users from accessing specified URLs or URLs that contain specified keywords in HTTP session only, but HTTPS session is not supported. Enter up to four URLs and up to six keywords.

Blocking by destination IP address

Check this box to prevent users from accessing specified IP addresses. Enter up to four IP addresses.

Blocking by Services

Check this box to prevent users from accessing specified Internet services, such as FTP or telnet. (You can block up to three applications per policy.) From the Applications list, click the application that you want to block. Then click the right-arrow button to move the application to the Blocked List. To remove an application from the Blocked List, click it and then click the button left-arrow button.

Application List:

• DNS(53 - 53)

• FTP(21 - 21)

• HTTP(80 - 80)

• HTTPS(443 - 443)

• TFTP(69 - 69)

• IMAP(143 - 143)

• NNTP(119 - 119)

• POP3(110 - 110)

• SMTP(25 - 25)

• SNMP(161 - 161)

• TELNET(23 - 23)

Network Setup > Firewall > IPV4 > Inbound Access Control

Field

Description

Inbound Access Control

Configure rules controlling your users' access from the Internet (WAN to LAN).

Add Entry

Click the Add Entry button to create another Advanced Firewall Policy entry

Advanced Firewall Policy List

- Policy Name

Show the entry of Policy Name

- Status

Show the entry of Status

- IN Interface(WAN)

Show the entry of IN Interface(WAN)

- OUT Interface(LAN)

Show the entry of OUT Interface(LAN)

- Priority

Show the entry of Priority

Rule Name

User specified rule name. Up to 31 characters are allowed to key-in.

Status

Enabled or disabled this rule entry.

IN Interface(WAN)

Select WAN interface to apply for a rule. ALL WAN means the IN Interface for this rule applied for all WAN interfaces.

OUT Interface(LAN)

Select LAN interface to apply for a rule. ALL LAN means the OUT Interface for this rule applied for all LAN interfaces.

Source IP Address

Matches the Source IP address to which packets are addressed to this rule.

Source Subnet Mask

Defines the source IP address wildcard mask. Masks specify which bits are used and which bits are ignored. A mask of 255.255.255.255 indicates that all the bits are important. A mask of 0.0.0.0 indicates that all the bits will be ignored. Therefore, if an Source IP Address is specified but source subnet mask specified to 0.0.0.0, the rule will regards it as 0.0.0.0/0 (all) address

Destination IP Address

Matches the destination IP address to which packets are addressed to this rule.

Destination Subnet Mask

Defines the destination IP address wildcard mask. Masks specify which bits are used and which bits are ignored. A mask of 255.255.255.255 indicates that all the bits are important. A mask of 0.0.0.0 indicates that all the bits will be ignored. Therefore, if an destination IP address is specified but destination subnet mask specified to 0.0.0.0, the rule will regards it as 0.0.0.0/0 (all) address.

Protocol

Specify the interested protocol for this rule. Default is "Any", which, means protocol filed filter will be disabled and all kind of protocol are going to inspect. Beside, user can specify UDP, TCP, or ICMP protocol

Source Port

Defines the TCP/UDP source port that this rule to match. “Any” means port field will not be inspected. “Single” means a port is specified. “Range” means that port range are specified.

Destination Port

Defines the TCP/UDP destination port that this rule to match. “Any” means port field will not be inspected. “Single” means a port is specified. “Range” means that port range are specified

Action

Deny or Permit the traffic associated with this rule.

Schedule

Selective week day schedule that this rule is going to apply.

Times

Specified time period that this rule is going to apply.

Network Setup > PPPoE Relay

Field

Description

Add Entry

Click the Add Entry button to create another PPPoE Relay

PPPoE Relay list

- Wan option

Show the entry of wan option.

- Lan option

Show the entry of lan option.

- PPPoE Relay

Show the entry of status.

PPPoE Relay

Enable or Disable PPPoE Relay

WAN Interface

Select the WAN Interface for this rule

LAN Interface

Select the LAN Interface for this rule

Network Setup > DDNS

Field

Description

DDNS

Dynamic DNS (DDNS) is an Internet service that allows routers with varying public IP addresses to be located using Internet domain names. If your ISP has not provided you with a static IP, and your WAN connection is configured to use DHCP to get an IP address dynamically, then DDNS allows you to have a virtual static address for your website. To use DDNS, you must setup an account with a DDNS provider such as DynDNS.com or TZO.com Use the DDNS page to activate your service on the router.

DynDNS.org

DynDNS.org

You must sign up for an account with DynDNS.org before you can use this service.

User Name

Enter the user name from DynDNS.org.

Password

Enter the password from DynDNS.org.

Host Name

Enter your host name. This should be in the format of name.dyndns.org.

System

Select the DynDNS service you use: Dynamic, Static, or Custom.

Mail Exchange (Optional)

Enter the address of your mail exchange server, so the email to your DynDNS address go to your mail server.

Mail Exchange (Backup MX)

This feature allows the mail exchange server to be a backup. To enable this feature, use the default setting, Enabled. To disable this feature, select Disabled. If you are not sure, which setting to select, use the default setting, Enabled.

Wildcard

This feature allows you to use a wildcard value in the DDNS address. For example, if your DDNS address is myplace.dyndns.org and you enable wildcard, then the x.myplace.dyndns.org will work as well (x is the wildcard). To enable wild cards, use the default setting, Enabled. To disable wildcard, select Disabled. If you are not sure which to select, use the default setting, Enabled.

Internet IP Address

Your current IP address.

Status

Your DDNS status.

Update

To manually trigger an update, click this button.

TZO.com

TZO

You must sign up for an account with TZO before you can use this service.

E-Mail Address

Enter the email address for your TZO account.

TZO Key

Enter the key for your TZO account.

Domain Name

Enter your host name. This should be in the format of name.dyndns.org.

Internet IP Address

Your current IP address.

Status

TZO DDNS status.

Update

To manually trigger an update, click this button.

Network Setup > DMZ

Network Setup > DMZ > Software DMZ

Field

Description

Software DMZ

A DMZ (Demarcation Zone or Demilitarized Zone) is a sub-network that is behind the firewall but that is open to the public. By placing your public services on a DMZ, you can add an additional layer of security to the LAN. The public can connect to the services on the DMZ but cannot penetrate the LAN. You should configure your DMZ to include any hosts that must be exposed to the WAN (such as web or email servers)

Add Entry

Click the Add Entry button to create a software DMZ entry

Status

Select enable to activate this entry, or disable to deactivate it.

Public IP

Input an public IP address that this DMZ server will expose to the Internet

Private IP

The Subnet Mask Server's private IP address behind LAN corresponding to the Public IP address

Network Setup > DMZ > Hardware DMZ

Field

Description

Hardware DMZ

This feature will use new LAN port 4 as can be used for DMZ purposes for public access to the customer's web and other servers that are accessible from the Internet. The rest LAN network ports will continue to be used for private internal traffic.Please note that this feature only supported while WAN in static or DHCP mode. Hardware DMZ site can't be applied for a VPN connection site.

Hardware DMZ

When select enabled, LAN port 4 will act as DMZ port, or it acts as a normal LAN port for private internal traffic.

Add Entry

Click the Add Entry button to create a hardware DMZ IP matching.

Hardware DMZ Details

Status

Select enable to activate this entry, or disable to deactivate it.

Public IP

Input an public IP address that equal to the server IP address that attached behind hardware DMZ port

Network Setup > IGMP

Field

Description

IGMP

Internet Group Management Protocol (IGMP) is a signaling protocol that supports IP multicasting for IPTV.

IGMP Proxy

Keep the default setting, Enabled, if you want to allow multicast traffic through the router for your multimedia application devices. Otherwise, select Disabled.

Support IGMP Version

Select the version you want to support, IGMP v1, IGMP v2,or IGMP v3. If you are not sure which version to select, keep the default setting, IGMP v2.

WAN Interface

Select WAN interface you want to forward, you can check Internet Setup to check its type. If you are not sure which WAN interface to select, keep the default setting [AUTO] to follow system default route interface.

Immediate Leave

Select Enabled, if you use IPTV applications and want to allow immediate channel swapping or flipping without lag or delays. Otherwise, use the default setting, Disabled.

Network Setup > UPnP

Field

Description

UPnP

UPnP (Universal Plug and Play) is a feature that allows for automatic discovery of devices that can communicate with the router.

UPnP

If you want to use UPnP, use the default setting, Enabled. Otherwise, select Disabled.

Allow Users to Configure

When this feature is enabled (the default setting), you can make manual changes while using the UPnP feature. Select Disabled, if you don't want to be able to make manual changes.

Keep UPnP Configurations After System Reboot

When this feature is enabled, the router saves UPnP configuration after a system reboot. The default is Disabled. When this feature is disabled, the router does not save UPnP configuration, but it does not remove the previous UPnP configuration.

Network Setup > CDP

Field

Description

CDP

Cisco Discovery Protocol (CDP) is a device discovery protocol that runs on all Cisco equipment. Each CDP-enabled device sends periodic messages to a multicast address and also listens to the periodic messages sent by others in order to learn about neighboring devices. Use the CDP page to choose the CDP settings for your network.

CDP Setting

CDP

You can enable CDP on all ports, disable CDP on all ports, or configure CDP per port. Cisco recommends the default setting, Per Port. Enabling CDP is not recommended on the WAN port because it is connected to an insecure network.

CDP Timer

Specify the interval at which successive CDP packets can be sent. Valid values are from 5-900. The default is 60.

CDP Hold Timer

Specify the amount of time that the information sent in the CDP packet is cached by the device that receives the CDP packet. Valid values are from 10-255. The default is 180.

Interface List

Checked the enable check box to enable the interface.

Network Setup > DNS Spoofing

Field

Description

Enable

Enable DNS spoofing

Add Entry

Add DNS spoofing entry

DNS Spoofing Add Entry Setting

Host Name

Enter one domain name field to spoofing.

IP Address

Enter one mapping IP address.

VPN module

The VPN module includes these pages:

- VPN > Site to Site IPSec VPN

- VPN > GRE Tunnel

- VPN > VPN Passthrough

- VPN > Cisco VPN Server

VPN > Site to Site IPSec VPN

VPN > Site to Site IPSec VPN > NAT Traversal

Field

Description

NAT Traversal

NAT Traversal

IPSec NAT Traversal can support detecting the presence of NAT. The detecting packet not only detects the presence of NAT between the two IKE peers, but also detects where the NAT is. The location of the NAT device is important, as the keepalives have to initiate from the peer behind the NAT. Please refer RFC3947.To enable this feature, choose Enabled. To disable this feature, choose Disabled.

VPN > Site to Site IPSec VPN > IKE Policy

Field

Description

IKE Policy

Add Entry

Click the Add Entry button to create another IKE policy.

List of IKE Policies

- Name

Show entry of name.

IKE Details

Select an entry from the List of IKE Policies, Details of IKE will show all information about IKE Policy.

General

Policy Name

Use a unique name which will be displayed in the list of VPN policies for the selection.

Exchange Mode

Choose the exchange mode based on your requirements for security and speed. Main: Choose this mode if you want higher security, but with a slower connection. Main Mode relies upon two-way key exchanges between the initiator and the receiver. The key-exchange process slows down the connection but increases security. Aggressive: Choose this mode if you want a faster connection, but with lowered security. In Aggressive Mode there are fewer key exchanges between the initiator and the receiver. Both sides exchange information even before there is a secure channel. This feature creates a faster connection but with less security than Main Mode.

Remote ID/Local ID

To set up remote and local identity, keep empty to remove identity setting. This can be an IP address (specified as dotted quad or as a Fully Qualified Domain Name, which will be resolved immediately) or as a Fully Qualified Domain Name itself (prefixed by ''@'' to signify that it should not be resolved)

IKE SA Parameters

Encryption Algorithm

The available encryption algorithms are, DES, 3DES, AES128, AES192, and AES256.

Authentication Algorithm

The available authentication algorithms are MD5 and SHA1.

Diffie-Hellman (DH) Group

Choose a DH group to set the strength of the algorithm in bits: Group 1 (768 bits) and Group 2 (1024bits).

Pre Shared Key

Enter an alpha-numeric key to be shared with IKE peer.

Enable Dead Peer (DPD) Detection

This function is not necessary for an IKE rule, but it will help to keep the connection alive during periods when there is no traffic.

DPD Interval

DPD packet is sent periodically in interval seconds during no data traffic.

DPD Timeout

The connection will be disconnected if there is no DPD response after DPD timeout. Unit is second.

Extended Authentication

XAUTH Client Enable

When this feature is enabled, the router can authenticate users from an external authentication server such as a RADIUS server. Enable this function only if the router is connected to a XAUTH server.

User Name/Password

Enter the credentials that the router uses to connect to the XAUTH server.

VPN > Site to Site IPSec VPN > IPSec Policy

Field

Description

IPSec Policy

A VPN policy contains IPSec Security Association parameters, which define the connection type and key type. Click the Add Entry button to add another VPN policy. To edit an existing policy, click the pencil icon.

Add Entry

Click the Add Entry button to create another IPSec policy.

List of VPN Policies

- Enable

Select the enable check box to enable the VPN entry.

- Number

Show Entry of number.

- NAME

Show Entry of name

VPN Details

Select one entry of List of VPN Policies, Details of VPN will show all information about VPN Policy.

General

Enable

Check to Enable IPSec Policy.

Policy Number

Enter an identification number for the policy.

Policy Name

Enter a unique name to be used to bring up the tunnel.

Policy Type

Choose Auto Policy or Manual Policy. The Auto Policy uses the IKE protocol to negotiate random keys for more security. You also must set an IKE policy on the Site to Site IPSec VPN > IKE Policy page The Manual Policy does not use IKE, which makes this policy more simple, but less secure.

Remote Endpoint

Choose how you want to identify the remote gateway for this site-to-site VPN tunnel. Choose IP Address to enter an IP address, choose FQDN to enter a Fully Qualified Domain Name, or choose Any (available only for an Auto Policy). Be aware that an FQDN requires that the router can connect to a DNS server to resolve the address before establishing the VPN tunnel.

Encryption Algorithm

Choose DES, 3DES, AES128, AES192, or AES256.

Integrity Algorithm

Choose MD5 or SHA1.

WAN Interface Name

Choose System Default Route, Ether_WAN1, USB_Modem.

Auto Policy Parameters

PFS

When used in the memo Perfect Forward Secrecy (PFS) refers to the notion that compromise of a single key will permit access to only data protected by a single key. For PFS to exist the key used to protect transmission of data MUST NOT be used to derive any additional keys, and if the key used to protect transmission of data was derived from some other keying material, that material MUST NOT be used to derive any more keys.

SA Lifetime

Enter the IPSec SA life time in seconds. The default value is 7800, which is 130 minutes.

Local Traffic Selection

Local IP

Choose the type of identifier that you want to use (IP Address or IP Address and Subnet Mask) for the local group that is allowed to pass through this tunnel then enter the identifier(s).

IP Address

Enter the IP Address.

Subnet Mask

Remote Traffic Selection

Remote IP

Choose the type of identifier that you want to use (IP Address or IP Address and Subnet Mask) for the local group that is allowed to pass through this tunnel then enter the identifier(s).

IP Address

Enter the IP Address.

Select IKE Policy

Choose an IKE policy to associate with this IPSec Policy. To view all IKE policies in a table, click the View IKE Table button.

VPN > GRE Tunnel

Field

Description

GRE Tunnel

Generic Routing Encapsulation (GRE) is a tunneling protocol developed by Cisco that can encapsulate a wide variety of network layer protocol packet type inside IP tunnels, creating a virtual point-to-point link to Cisco routers at remote points over an IP Internet network.

Add Entry

Click the Add Entry button to create another GRE tunnel

Summary GRE Tunnel

- Number

Displayed here is the number which you selected.

- Status

Displayed here is the status of the tunnel.

Tunnel Name

Displayed here is the name of the tunnel.

GRE Details

Select one GRE tunnel of Summary GRE tunnel, GRE Details will show all Information about GRE. (like Status, Checksum, Sequence, Key, Key Value, Tunnel Name, Destination IP or HostName and Remote IP Address / Subnet mask)

GRE IP Tunnel

Tunnel Number

Choose an identification number for this tunnel.

Tunnel Name

Enter a name to describe this tunnel.

Enable

Check the box to enable the tunnel, or uncheck the box to disable the tunnel.

Checksum

Choose Input, Output, Both, or None. Input requires that all inbound packets have the correct checksum. Output requires the checksums for outbound packets. Both require the checksum for all inbound and outbound packets. The default is None.

Sequence

Choose None, Both, Input, or Output. Output requires a sequence number for outbound packets. Input requires a sequence number for inbound packets. Both require a sequence number for inbound and outbound packets. The default is None.

If sequence number check is set as Input or Both in receiver side, when sender side GRE session restart, the connection will be resumed after the sequence number reach the amount that record in previous session.

Key

Choose Input, Output, Both, or None. Output requires a key for outbound packets. Input requires a key for inbound packets. Both require a key for inbound and outbound packets. The default is None.

Key Value

If you chose Input, Output, or Both for the Key, specify the key by entering a number between 1 and 4294967295.

WAN Interface Name

Choose the WAN interface that is used to create the GRE Tunnel with the remote host.

Destination IP or HostName

Enter the Destination IP is the address of the remote network or host to which you want to build a tunnel with it.

Remote IP Address/Subnet Mask

Select the Remote IP Address/Subnet Mask for the remote host. You can use the below Add/Delete button to add/delete the pair

Modify Remote IP Address/Subnet Mask

You can input the pair of Remote IP Address and Subnet mask in this field. And then use the Add button to add it into the list of Remote IP Address/Subnet Mask. The following is example for this field: 192.168.2.0/24 or 192.168.3.0/32.

VPN > VPN Passthrough

Field

Description

VPN Passthrough

Configure IPSec passthrough if there are devices behind the router that need to set up IPSec tunnels independently, for example, to connect to another router on the WAN.

IPSec Passthrough

Internet Protocol Security (IPSec) is a suite of protocols used to implement secure exchange of packets at the IP layer. IPSec Passthrough is enabled by default. To disable IPSec Passthrough, select Disabled.

PPTP Passthrough

Point-to-Point Tunneling Protocol (PPTP) allows the Point-to-Point Protocol (PPP) to be tunneled through an IP network. PPTP Pass-Through is enabled by default. To disable PPTP Passthrough, select Disabled.

L2TP Passthrough

Layer 2 Tunneling Protocol is the method used to enable Point-to-Point sessions via the Internet on the Layer 2 level. L2TP Pass-Through is enabled by default. To disable L2TP Passthrough, select Disabled.

VPN > Cisco VPN Server

VPN > Cisco VPN Server > Group

Field

Description

Group

The Cisco VPN Server allows mobile users to access Intranet resource via an encrypted (IPSec) VPN tunnel by Cisco Systems VPN Client. The default values of IKE phase 1 and 2 are accepted by Cisco VPN client. Due to system restriction, “Cisco VPN Server” and “Site to Site VPN” are mutually exclusive.

Enable

Click Enable to activate the VPN server. The default is Disable.

Enabling the VPN Server will deactivate any site-to-site VPN tunnels that have been defined.

Identify

Group Name

Cisco VPN Group name used as an identifier for the VPN server. This name must match the group name specified the VPN Client profile. The length can contain up to 32 characters.

Password

Cisco VPN Group password. This password must match the group password specified the VPN Client profile. The length can contain up to 32 characters.

IKE Phase 1

Exchange Mode

Aggressive mode is applied by default and cannot be changed. This mode is used for negotiating phase one ISAKMP Security Associations (SAs) when using preshared keys for authentication.

ESP Algorithm

Enter an encryption algorithm for the ISAKMP SA.Choices are AES, DES, and 3DES. The default is AES.

AH Algorithm

Hash algorithm for the ISAKMP SA. Choices are MD5 and SHA1. The default is MD5.

Auth Method

Method used to authenticate the remote user. Choices are PSK or PSK+XAUTH. If PSK is selected, then the client will be authenticated if it specifies the correct group name and password. If PSK+XAUTH is selected, then an additional username and password is required.

DH Group

Diffie-Hellman group options. Only 2 [modp 1024], The default is 2 [modp 1024]

IKE Phase 2

PFS Group

Diffie-Hellman exponentiation group. Choices are: 1 [modp 768], 2 [modp 1024], 5 [modp 1536], 14 [modp 2048], or 15 [modp 3072].

SA Life Time

Defines how long an IPSec SA (security association) will be used. The default is 30 minutes.

Mode Configuration

Starting IP Address

Starting IP address of the range of addresses that are assigned to the remote client. This range must not be in the same subnet as any VLAN.

Subnet Mask

Subnet mask for the address range assigned to remote clients.

DNS1

Primary DNS server to be used by remote clients.

DNS2

Secondary DNS server to be used by remote clients.

WINS1

Primary WINS server to be used by remote clients.

WINS2

Secondary WINS server to be used by remote clients

Banner

Message displayed to the remote user after they log on. The banner allows up to 500 printable ASCII characters on 1 line.

VPN > Cisco VPN Server > User

Field

Description

VPN Server Users

The Users page contains a list of usernames and passwords that can login to the Cisco VPN Server. Up to 15 unique users can be defined

Add Entry

Add User

List of VPN Server Users

List all the VPN users

User Account

Username

Username to be provided by the VPN client when using PSK+XAUTH as the authentication method.

Password

Password to be provided by the VPN client when using PSK+XAUTH as the authentication method.

Confirm password

The contents of this field must match the Password field.

Administration module

The Administration module includes these pages:

- Administration > Web Access Management

- Administration > Remote Support

- Administration > Remote Management

- Administration > Time Setup

- Administration > Certificate Management

- Administration > User Management

- Administration > User Privilege Control

- Administration > Log

- Administration > Factory Defaults

- Administration > Firmware Upgrade

- Administration > Backup & Restore

- Administration > Reboot

- Administration > Switch Setting

- Administration > Status

Administration > Web Access Management

Field

Description

Web Access Management

Allows you to change the Router's access settings.

Web Utility Access

To access this web utility, you can use no security by selecting HTTP or security by selecting HTTPS. If you select HTTPS, be aware that you will need to include https in the address when you connect to the utility. Refer to the following example: https://xxx.xxx.xxx.xxx (the x's represent the Gateway's Internet IP address).

Web Utility Access via Wireless

This feature allows the administrator to access web utility from a wireless device.

Login Banner

Banner Text

Input the Banner Text, the 1024 character left limitation

Remote Access

Remote Management

This feature allows you to manage your Gateway from a remote location, via the Internet. If you enable this option and have not changed the router password from the default value, you will be prompted to change the password for security purposes.

Web Utility Access

To access this web utility, you can have no security HTTP or security HTTPS. For HTTPS, enter https://xxx.xxx.xxx.xxx (the x's represent the Gateway's Internet IP address) in your web browser's Address field.

Remote Upgrade

If enabled, the router firmware can be upgraded from Internet.

Allowed Remote IP Address

If you want to access the Router from any external IP address, select Any IP Address. If you want to specify an external IP address or range of IP addresses, then select the second option and complete the fields provided.

Remote Management Port

Enter the port number that will be open to outside access

Administration > Remote Support

Field

Description

Remote Support Access

Collect Device Status Information

Click this button will collect system configuration and useful routing information that can help to debug this system.

Enable Remote Support

Turn on remote debug shell.

Access Port

The debug shell's port number. Default is port 22.

Administration > Remote Management

Administration > Remote Management > TR-069

Field

Description

TR-069

Some service providers can automatically provision your customer premises equipment from a central server. Use the TR-069 page to set up communication with an Auto Configuration Server (ACS).

Status

Click Enabled to allow auto-configuration of your router from a central server. Otherwise, click Disabled.

ACS URL

Enter the address and port of the ACS server. The format should be http(s)://xxx.xxx.xxx.xxx:port or xxx.xxx.xxx.xxx:port or http(s)://xxx.xxx.xxx.xxx:port/zzzz or xxx.xxx.xxx.xxx/zzz. The X's represent the IP address or domain name. The Z’s represent the URL location. After the colon, enter the port number.

ACS UserName

The default username is OUI-Serial Number; this value should be the same as configured at ACS side and must be filled.

ACS Password

This value should be the same as configured at ACS side and must be filled.

Connection Request Port

This port receives the Connection Request notification from the ACS

Connection Request Username

This value should be the same as configured at ACS side.

Connection Request Password

This value should be the same as configured at ACS side.

Periodic Inform Enable

Choose Enabled to allow the router to periodically initiate connections to the ACS. Otherwise, choose Disabled.

Periodic Inform Interval

Specify the interval (in seconds) at which the router will initiate connections to the ACS. The default value is 86400 seconds, which is 24 hours.

Binding with Loopback Interface

To check the Binding with Loopback Interface box and select a Loopback Interface to bind IP of the interface with TR-069 Connection request URL. The default is unchecked, which is to bind default WAN IP with Connection request URL.

Request Download

Click the Apply button if you want to immediately initiate a connection to the ACS. The ACS may call the Download RPC when it receives the request.

Provisioning Code

This value could be used by ACS to determine service provider-specific customization and provisioning parameters.

Administration > Remote Management > SNMP

Field

Description

SNMP

Simple Network Management Protocol (SNMP) is a popular network monitoring and management protocol that lets you monitor and manage your network from an SNMP manager. SNMP provides a remote means to monitor and control network devices, and to manage configurations, statistics collection, performance, and security.

SNMP Setting

SNMP

To enable SNMP identification, click Enabled . To disable SNMP, click Disabled .

Trusted IP

Choose Any to allow access from any IP address (not recommended) or enter the IP address and subnet mask of a single SNMP manager or trap agent that can access this router via SNMP.

Get Community

Enter the password that allows read-only access to the Gateway's SNMP information.

Set Community

Enter the password that allows read/write access to the Gateway's SNMP information.

SNMPV3

To enable SNMPV3 function, click Enabled . To disable SNMPV3, click Disabled .

R/W User

Enter the user name for SNMPV3

Auth-Protocol

Choose SNMPV3 auth protocol, available protocol is “HMAC-MD5” and “HMAC-SHA”

Auth-Password

Enter password for Auth check.

PrivProtocol

Authentication is performed by using a users privKey to encrypt the data portion the message being sent.

Privacy Password

Enter the privKey for PrivProtocol to use.

SNMP Trap

To enable SNMP Trap, click Enabled. To disable SNMP Trap, click Disabled. SNMP Trap can be enabled only when SNMP is enabled

Trap Server

Enter the IP address that trap will be sent to.

Trap Community

Enter the password that allow read access to the SNMP Trap message.

Trap User

Enter the user name for SNMPV3 Trap

Trap Auth- Protocol

Choose SNMPV3 Trap auth protocol, available protocol is "HMAC-MD5" and “HMAC-SHA”.

Trap Auth- Password

Enter SNMPV3 Trap password for Auth check

Trap PrivProtocol

SNMPV3 Trap authentication is performed by using a users privKey to encrypt the data portion the message being sent.

Trap Privacy Password

Enter SNMPV3 Trap privKey for PrivProtocol to use.

Administration > Remote Management > Local TFTP

Field

Description

Local TFTP Control

TFTP

Control TFTP enabled or disabled. Default Enabled.

Get Remote File

URL

This shows where can get remote file.

Save As

Specify the file name to save

Session Timeout

Maximum time allowed for a connection session. A connection timeout for HTTP and FTP session will be 3 seconds, TFTP will be 1 seconds. For HTTP and FTP, a TCP reset response message will terminate a session.

Retry Sessions

Specify how many sessions are going to retry if transient problem occurred in a session

Status

The status of processing get remote file

File List

- Name

This is the name of local file.

- Size

This is the size of local file.

Administration > Time Setup

Field

Description

Time Zone

Setup the time zone and configure the system time by synchronizing with time server (NTP) or set time manually (Manual Setting).

Time Zone

Select the time zone in which your network functions from this drop-down menu.Time zone is a region of the earth that has uniform standard time, usually referred to as the local time.

NTP

Time Server Address

If you want to use the device's default Network Time Protocol (NTP) server, use the default setting, Auto. If you want to specify the NTP server, select Manual, and enter the URL or IP address of the NTP server you want to use.

Resync Timer

The timer controls how often the Device resyncs with the NTP server. Enter the number of seconds you want the interval to be, or use the default setting, 3600 seconds.

Enable Daylight Saving

Select this option if you want the device to automatically adjust for daylight saving time. This option is enabled by default

Manual Setting

Date

date in format “Year/Month/Day”

Time

time in format “Hour:Min:Sec”

Auto Recovery After Reboot

Auto Recovery After Reboot

When this feature is enabled, the device will recover system time after system reboot.

Administration > Certificate Management

Field

Description

Certificate Management

To support uploading certificate authority file through WEB GUI for TR069 and Provision. Up to 3 certificate authority files can be uploaded for T069 and 1 certificate authority file can be uploaded for Provision.

TR069 - Root CA File List

Enabled

After uploading certificate authority file, click the check box to allow TR069 using the file in certification. Deselect all check boxes to disable all certificate authority file used by TR069. Please note, only one certificate authority file can be selected in the same time.

CA Name

To set the certificate authority file name in the system.

Select Certificate

To select a certificate authority file in client PC, and click Upload button to uploading the file. After uploading, you can click Enabled check box or click icon to delete the file.

Provision File List

Enabled

After uploading certificate authority file, click the check box to allow Provision using the file in certification. Deselect all check boxes to disable all certificate authority file used by Provision.

CA Name

To set the certificate authority file name in the system.

Select Certificate

To select a certificate authority file in client PC, and click Upload button to uploading the file. After uploading, you can click Enabled check box or client icon to delete the file.

Administration > User Management

Administration > User Management > Password Complexity Settings

Field

Description

Password Complexity Settings

Password Complexity

Click Enabled to activate the User Password Complexity. The default is Disabled.

Password Complexity check Level:

- Low - Too Short Password

- Low - Passwords cannot be repeated consecutively for three times

- Low - Weak Password, use letters & numbers.

- Medium - Medium Password, Use special charecters

- High - Strong Password

- Password is the same as username.

Administration > User Management > User List

Field

Description

User List

Use the User List page to manage the users who have access to the router configuration utility. There are two default accounts. The account with the default username of admin has administrator-level access. The account with the default username of cisco has guest-level access.

User Account

Username

This is the name to login router.

Level

This shows user's level.

User List

Username

Enter a new Username. The two default usernames cannot be changed.

Old Password

To ensure the device's security, you will be asked for your old password when you want to change the password. The default administrator password is admin. The default guest password is cisco. Cisco strongly recommends changing the password.

New Password

To ensure the device's security or WRP500's security, you will be asked for your password when you access the device's configuration utility. The default administrator password is admin. The default guest password is cisco. Cisco strongly recommends changing the password

Confirm New Password

Enter the new password again to confirm.

Level

The level of permission for this user: Admin or User. Admin has access to all settings as specified on the Privilege Control page. User has read-only access.

Administration > User Privilege Control

The privilege control provides three access types for all webpages: Read/Write, Read Only and Hidden. The Read/Write means to allow view and configure the items of the webpage. The Read Only means only allow view the webpage. The Hidden means the no any hyperlink to the webpage.

Administration > Log

Administration > Log > Log Setting

Field

Description

Local

Local

To save log message in memory of router, after reboot, all the logs will disappear.

Log size

Up limit to save log message in memory, the allowed range is 128~1024KB.

USB

USB

To save log message in external USB storage, if no USB storage plugs in, only “USB disconnect” shows. If USB storage is connected to, user can set

File Name

Filename to be saved into USB disk

Log size

Up limit to save log message in USB storage, the allowed range is 1~512MB

Syslog Server

Syslog Server

Send out log message to remote syslog server.

IP Address

Enter IP address of remote syslog server

Port

Enter port number that syslog server listen on. Port 514 is chosen by default.

E-Mail

E-Mail

Send out log message to specific E-Mail address.

Sender

Specify sender's E-Mail address.

Receiver

Specify receiver's E-Mail address.

SMTP Server

Enter mail server address.

SMTP Port

Enter port number that mail server listen on. Port 25 is chosen by default.

Subject

Specify mail subject to send log.

Number of logs

Enter a number to specify how many logs are collected in an E-Mail.

Interval

Enter a time interval to force send out E-Mail if the amount of logs doesn't reach Number of logs

User Name

Enter a user name for mail server authentication.

Password

Enter a password for mail server authentication.

Administration > Log > Log Module

Field

Description

Log Module Settings

Status

To enable the collection of activity logs, select Enabled, and then click Submit. With logging enabled, you can choose to view temporary logs. Click the Disabled radio button to disable this function

Log

This drop-down list becomes available if you enable logging and choose log target to decide where the log save to.

- Local

Save log to system memory

- USB

Save log to USB disk, only work when USB disk is plugged in.

- E-Mail

Send log through E-Mail, please setup E-Mail related information in Log Setting page.

- Syslog Server

Send log to specific log server, please setup log server address in Log Setting page

Administration > Log > Log Viewer

Field

Description

Log Viewer

Allow user to see, download or clean log message save in system memory

Download All Log

Click to download log message in a file to local PC

Clear Log

Click to clean all log message saved in memory.

Display

Choose module to see related log message.

Filter

To filter log message with specific pattern.

Administration > Log > Firewall Log

Field

Description

Firewall Log

Firewall Log provides a functionality that can log certain specified traffic according to the current system firewall, such as SPI and DoS attacking. The traffic that matches the specified firewall rules will be logged. Firewall Log configuration page is shown as below. The description of each configured fields are explained as below.

Firewall Log Settings

Firewall Log

Enable or disable firewall logging.

Log Level

Level of logging by using the specified syslog level:

- 0 Emergency: system is unusable

- 1 Alert: action must be taken immediately

- 2 Critical: critical conditions

- 3 Error: error conditions

- 4 Warning: warning conditions

- 5 Notice: normal but significant condition

- 6 Info: Info messages

- 7 Debug: debug-level messages

Log Category

Select which firewall module that is going to be logged and set how many events that generate one log.

Administration > Factory Defaults

Field

Description

Factory Defaults

The Factory Defaults screen allows you to restore the Router's Configuration to its Router and/or voice factory default settings.

Note Restoring the voice defaults may require your login (the default user name and password are admin). If the defaults do not work, contact your ITSP for more information.

Factory Defaults

Restore Router Factory Defaults

To reset the data (router) settings to the default values, select Yes, then click Submit. Any custom data (router) settings you have saved will be lost when the default settings are restored.

Restore Voice Factory Defaults

To reset the voice settings to the default values, select Yes, then click Submit. Any custom voice settings you have saved will be lost when the default settings are restored.

Administration > Firmware Upgrade

Field

Description

Firmware Upgrade

The Firmware Upgrade screen allows you to upgrade the Router's firmware. You do not need to upgrade the firmware unless you are experiencing problems with the Router or the new firmware has a feature you want to use.

Before upgrading the firmware, download the Router's firmware upgrade file from the Cisco website, www.cisco.com. Then extract the file.

Note The Router may lose the settings you have customized. Before you upgrade its firmware, write down all of your custom settings. After you upgrade the firmware, you will have to re-enter all of your configuration settings.

Firmware Upgrade Settings

Please select a file to upgrade.

In the field provided, enter the name of the extracted firmware upgrade file, or click the Browse button to find this file.

Upgrade

After you have selected the appropriate file, click this button, and follow the on-screen instructions.

Administration > Backup & Restore

Administration > Backup & Restore > Default Configuration

Field

Description

Default Configuration

Specifies the Default Configuration settings.

Load Service Provider Default Configuration

Select Yes to load Service Provider default configure when do system factory default, select No to load Cisco factory default.

Administration > Backup & Restore > Backup Configuration

Field

Description

Backup Configuration

To back up the router's configuration settings

Backup

To back up the Router's configuration settings, click this button and follow the on-screen instructions.

Administration > Backup & Restore > Restore Configuration

Field

Description

Restore Configuration

To backup current configuration in case you need to reset the router back to its factory default settings.

Please select a file to restore

To restore the Router's configuration settings, click this button and follow the on-screen instructions. (You must have previously backed up the Router's configuration settings.)

Administration > Reboot

Click Reboot to power cycle the router.

Administration > Switch Setting

Administration > Switch Setting > Port Status

Field

Description

Port Status

Active/Inactive switch wire port. When deactivated, this port cannot do any network function until it is reactivated.

Port Status Setting

Interface

The wire physical port that support on/off by administrator, don't include wireless or pvc interface.

Enabled

Click to allow network traffic input/output from this physical port. When administrator unclicks this port, LED will be off and traffic cannot pass.

Administration > Switch Setting > Bind MAC to Port

Field

Description

Bind MAC to Port

Enable this function will bind the assigned mac address to one of the LAN ports, and only allow this mac address can access this assigned LAN port but not others port.

Bind MAC to Port Setting

Adding MAC address

Administrator add new entry to allow network traffic which source MAC come from which physical wire port.

- LAN Port

The physical wire port that support will bind to this mac address, not include wireless or PVC port.

- MAC Address

DUT will allow network traffic which source MACs (amount of 16) to match this setting.

- Add

Button that add this bind LAN Port/MAC address into filter table.

Enable Bind MAC to LAN Port 1

All MAC address entries that LAN Port 1 is relative

- Enable

click button to on/off Bind MAC address to LAN Port 1 function.

- MAC Address

address lists that administrator setting at LAN Port 1.

Enable Bind MAC to LAN Port 2

All MAC address entries that LAN Port 2 is relative.

- Enable

click button to on/off Bind MAC address to LAN Port 2 function.

- MAC Address

address lists that administrator setting at LAN Port 2.

Enable Bind MAC to LAN Port 3

All MAC address entries that LAN Port 3 is relative.

- Enable

click button to on/off Bind MAC address to LAN Port 3 function.

- MAC Address

address lists that administrator setting at LAN Port 3.

Enable Bind MAC to LAN Port 4

All MAC address entries that LAN Port 4 is relative.

- Enable

click button to on/off Bind MAC address to LAN Port 4 function.

- MAC Address

address lists that administrator setting at LAN Port 4.

Administration > Status

Field

Description

Status

CPU

This shows CPU's MIPS, Loads and Uptime

- Loads

This shows CPU's Loads.

- Uptime

This shows CPU's Uptime.

Memory

This shows Memory's Total size(%), Free size(%), Used size(%), Buffer size(%), Cached size(%), active size and inactive size(%).

- Total

This shows Memory's total size(%).

- Free

This shows Memory's free size(%).

- Used

This shows Memory's used size(%).

- Buffers

This shows Memory's buffer size(%).

- Cached

This shows Memory's cached size(%).

- Active

This shows Memory's active size(%).

- Inactive

This shows Memory's inactive size(%).

Troubleshooting

This appendix provides solutions to problems that may occur during the installation and operation of the WRP500s.

Note If you can't find an answer here, visit Cisco Community Central > Small Business Support Community at the following URL:

www.myciscocommunity.com/community/smallbizsupport/voiceandconferencing/ata

Q. I want to access the Configuration Utility, but the address I entered did not work.

If the device has ever been configured to allow access from the WAN interface, use the Interactive Voice Response Menu to find the Internet IP address. Follow these steps:

1. Use a telephone that is connected to the Phone 1 port of the WRP500.

2. Press **** (in other words, press the star key four times).

3. After the greeting plays, press 110# .

4. Write down the IP address as it is announced.

5. Open a web browser on a networked computer.

6. Start Internet Explorer and enter the IP address of the WRP500.

If the device has never been configured (that is, it still has the factory default configuration):

1. Connect PC to the LAN port. The PC should obtain the IP address through DHCP; the gateway is the IP address of the WRP500. For example, if the PC receives IP address 192.168.15.100, the WRP500 IP address is 192.168.15.1.

2. Enter web page.

3. Use default account admin: admin to login.

4. Navigate to Administration > Web Access Management.

5. Set remote management to enabled and remote management port to 80 .

6. Follow the steps from the previous Answer (if the device has ever been configured) to access the device web page through the WAN interface.

Q. I am trying to access the Configuration Utility, but I do not see the login screen. Instead, I see a 404 Forbidden screen.

If you are using Windows Explorer, perform the following steps until you see the Configuration Utility login screen. (Mozilla requires similar steps.)

1. Click File. Make sure Work Offline is NOT checked.

2. Press CTRL + F5 . This is a hard refresh, which forces Windows Explorer to load new web pages instead of cached ones.

3. Click Tools.

4. Click Internet Options.

5. Click the Security tab.

6. Click the Default level button.

7. Ensure that the security level is Medium or lower.

8. Click the OK button.

Q. How do I save the voice configuration for my WRP500?

1. Log in as admin.

2. Navigate to Administration > Backup & Restore > Backup Configuration.

3. Click the Backup button. The configuration is downloaded to your PC.

4. This .cfg file is helpful to provide to the support team when you have a problem or technical question.

Q. How do I debug the WRP500? Is there a syslog?

The WRP500 provides the option to send messages to both a syslog and debug server. The ports can be configured (by default, the port is 514).

1. Make sure you do not have a firewall running on your computer that can block port 514.

2. Start Internet Explorer, connect to the Configuration Utility.

3. Login as admin.The default username and password are both admin .

4. Under the Voice > System menu, set Syslog Server and Debug Server as the IP address and port number of your syslog server. Note that this address has to be reachable from the WRP500. For example, if the WRP500 is at 192.168.15.1, reachable addresses are in the range of 192.168.15.x, for example 192.168.15.100:514.

5. Set Debug level to 3. You do not need to change the value of the syslog server parameter.

6. Set Debug Option to dbg.all .

7. To capture SIP signaling messages, under the Voice > Line page, set SIP Debug Option to Full . The file output is syslog.<portnum>.log (for the default port setting, syslog.514.log).

Q. How do I access the WRP500 if I forget my password?

By default, the User and Admin accounts have no password. If the ITSP sets the password for either account and you do not know it, you need to contact the ITSP.

If the password for the user account was configured after you received the WRP500, you can reset the device to the user factory default, which preserves any provisioning that the ITSP completed.

If the Admin account needs to be reset, you have to perform a full factory reset, which also erases any provisioning.

To reset the WRP500 to the factory defaults, perform the following steps:

1. Connect an analog phone to the WRP500 and access the IVR by pressing **** .

2. Press the appropriate code to reset the unit:

– Press 73738# to perform a full reset of the unit to the factory default settings. The Admin account password will be reset to the default of blank.

3. Press 1 to confirm the operation, or press * to cancel the operation.

4. Log in to the unit by using the User or Admin account without a password.

5. Reconfigure the unit as necessary.

Q. The WRP500 is behind a NAT device or firewall. I am unable to make a call or I am only receiving a one-way connection. What should I do?

Complete the following steps:

1. Configure your router to port forward TCP port 80 to the IP address of the WRP500. You should use a static IP address. (For help with port forwarding, consult the documentation for the NAT device or firewall.)

2. On the Line tab of the Configuration Utility, change the value of Nat Mapping Enable to yes . On the SIP tab, change Substitute VIA Addr to yes , and the EXT IP parameter to the IP address of your router.

3. Make sure you are not blocking the UDP PORT 5060,5061 and port for UDP packets in the range of 16384-16482.

4. Disable SPI if this feature is provided by your firewall.

5. Identify the SIP server to which the WRP500 is registering. If it supports NAT, using the Outbound Proxy parameter.

6. Add a STUN server to allow traversal of UDP packets through the NAT device. On the SIP tab of the Configuration Utility, set STUN Enable to yes , and enter the IP address of the STUN server in STUN Server .

STUN (Simple Traversal of UDP through NATs) is a protocol defined by RFC 3489. STUN allows a client behind a NAT device to find out its public address, the type of NAT it is behind, and the port associated on the Internet connection with a particular local port. This information is used to set up UDP communication between two hosts that are both behind NAT routers. Open source STUN software can be obtained at the following address:

http://www.voip-info.org/wiki-Open+Source+VOIP+Software

Note STUN does not work with a symmetric NAT router. Enable debug through syslog (see FAQ#10), and set STUN Test Enable to yes. The messages indicate whether you have symmetric NAT or not.

Q. My computer cannot connect to the Internet. What should I do?

Follow this procedure:

1. Ensure that the Unified Communications Platform is powered on. The Power/Sys LED should be solid green and not flashing.

2. If the Power LED is flashing, power off all of your network devices, including the modem, the Unified Communications Platform, and the computers.

3. Wait for 30 seconds.

4. Power on each device in the following order:

– Cable or DSL modem

– Unified Communications Platform

– Computer

5. Check the cable connections. The computer should be connected to one of the ports numbered 1-4 on the Unified Communications Platform. The modem must be connected to the WAN (Internet) port on the Unified Communications Platform. For those ADSL devices that do not come with a modem, connect the ADSL line (normally, the digital phone line) to the WAN (Internet) port on the Unified Communications Platform.

Q. The computer cannot connect wirelessly to the network. What should I do?

Make sure the wireless network name or SSID is the same on both the computer and the Unified Communications Platform. If you have enabled wireless security, make sure the same security method and key are used by both the computer and the Unified Communications Platform.

Q. I upgraded my firmware and now the Unified Communications Platform is not working properly. Why?

If the Unified Communications Platform is not working properly after an upgrade, you may need to perform a factory reset. To perform a factory reset, use a ball pen point or a paper clip to poke through the hole labeled reset on the side of the Unified Communications Platform.

Q. There is no dial tone, and the Phone 1 or 2 LED is not solid green. What should I do?

Follow this procedure:

1. Make sure the telephone is plugged into the appropriate port, Phone 1 or 2.

2. Disconnect and re-connect the RJ-11 telephone cable between the Unified Communications Platform and telephone.

3. Ensure your telephone is set to its tone setting (not pulse).

4. Ensure your network has an active Internet connection. Try to access the Internet, and check to see whether the Unified Communications Platform WAN LED flashes green.

a. If you do not have a connection, power off all of your network devices, including the modem, the Unified Communications Platform, and the computers.

b. Wait 30 seconds.

c. Power on each device in the following order:

– Cable or DSL modem

– Unified Communications Platform

– Computers and other devices

5. Verify your account information and confirm that the phone line is registered with your Internet Telephony Service Provider (ITSP).

Q. When I place an Internet phone call, words are dropped intermittently. Why?

Consider the following possible causes and solutions:

- Cordless phone

If you are using the Unified Communications Platform wireless function and a cordless phone, they may be using the same radio frequency and may interfere with each other. Move the cordless phone farther away from the Unified Communications Platform.

- 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 be sure to exit the program before you make Internet phone calls.

- Bandwidth

There may not be enough bandwidth available for your Internet phone call. You may want to test your bandwidth by using one of the bandwidth tests that are 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. The DSL telephone line does not fit into the Unified Communications Platform WAN (Internet) port. Why?

The Unified Communications Platform does not replace your modem. You still need your DSL modem in order to use the Unified Communications Platform. Connect the telephone line to the DSL modem, re-run the setup wizard, then follow the on-screen instructions.

Q. The 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 Unified Communications Platform, you need a cable/DSL modem and a high-speed Internet connection.

Q. The Unified Communications Platform does not have a coaxial port for the cable connection.

The Unified Communications Platform does not replace your modem. You still need your cable modem in order to use the Unified Communications Platform. Connect your cable connection to the cable modem, re-run the setup wizard, then follow the on-screen instructions.

Environmental Specifications for the WRP500

Device Dimensions

6.69 x 6.69 x 1.30 in. (170 x 170 x 33 mm)

(includes foot)

Unit Weight

15.52 oz (440g)

Power

External, switching 12 VDC 1.67A

Certification

FCC, CE, CB,IC, UL

Wi-Fi (802.11ac/b/g/n, WPA2, WMM, WMM Power Save, and WPS2.0)

Operating Temperature

32 to 104°F (0 to 40°C)

Storage Temperature

-22 to 140°F (-30 to 60°C)

Operating Humidity

5 to 95%, noncondensing

Storage Humidity

5 to 95%, noncondensing

Where to Go From Here

This appendix describes additional resources that are available to help you and your customer obtain the full benefits of the Cisco WRP500.

Support

Cisco Small Business Support Community

www.cisco.com/go/smallbizsupport

Online Technical Support and Downloads (Login Required)

www.cisco.com/support

Small Business Support Center (SBSC) Phone Support Contacts

www.cisco.com/en/US/support/tsd_cisco_ small_business_support_center_contacts.html

Cisco Small Business Support and Resources

www.cisco.com/go/smallbizhelp

Cisco Small Business Firmware Downloads

www.cisco.com/go/software

Product Documentation

Product Documentation for Cisco Small Business Voice Gateways and ATAs

http://www.cisco.com/c/en/us/support/unified-communications/small-business-voice-gateways-ata/tsd-products-support-series-home.html

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

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The following information is for FCC compliance of Class A devices: This equipment has been tested and found to comply with the limits for a Class A digital device, pursuant to part 15 of the FCC rules. These limits are designed to provide reasonable protection against harmful interference when the equipment is operated in a commercial environment. This equipment generates, uses, and can radiate radio-frequency energy and, if not installed and used in accordance with the instruction manual, may cause harmful interference to radio communications. Operation of this equipment in a residential area is likely to cause harmful interference, in which case users will be required to correct the interference at their own expense.

The following information is for FCC compliance of Class B devices: This equipment has been tested and found to comply with the limits for a Class B digital device, pursuant to part 15 of the FCC rules. These limits are designed to provide reasonable protection against harmful interference in a residential installation. This equipment generates, uses and can radiate radio frequency energy and, if not installed and used in accordance with the instructions, may cause harmful interference to radio communications. However, there is no guarantee that interference will not occur in a particular installation. If the equipment causes interference to radio or television reception, which can be determined by turning the equipment off and on, users are encouraged to try to correct the interference by using one or more of the following measures:

- Reorient or relocate the receiving antenna.

- Increase the separation between the equipment and receiver.

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- Consult the dealer or an experienced radio/TV technician for help.

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