Cisco WRP400-G1 - Wireless Router, WRP400, Small Business Pro WRP400 Administration Manual

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Cisco Small Business Pro WRP400

Wireless-G Broadband Router with 2 Phone Ports and Built-In Analog Telephone Adapter

ADMINISTRATION

GUIDE

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CCDE, CCSI, CCENT, Cisco Eos, Cisco HealthPresence, the Cisco logo, Cisco Lumin, Cisco Nexus, Cisco Nurse Connect, Cisco Stackpower,

Cisco StadiumVision, Cisco TelePresence, Cisco WebEx, DCE, and Welcome to the Human Network are trademarks; Changing the Way We Work, Live, Play, and Learn and Cisco Store are service marks; and Access Registrar, Aironet, AsyncOS, Bringing the Meeting To You, Catalyst, CCDA, CCDP, CCIE, CCIP, CCNA, CCNP, CCSP, CCVP, Cisco, the Cisco Certified Internetwork Expert logo, Cisco IOS, Cisco Press, Cisco Systems, Cisco Systems Capital, the Cisco Systems logo, Cisco Unity, Collaboration Without Limitation, EtherFast, EtherSwitch, Event Center, Fast Step, Follow Me Browsing, FormShare, GigaDrive, HomeLink, Internet Quotient, IOS, iPhone, iQuick Study, IronPort, the IronPort logo, LightStream, Linksys, MediaTone, MeetingPlace, MeetingPlace Chime Sound, MGX, Networkers, Networking Academy, Network Registrar, PCNow, PIX, PowerPanels, ProConnect, ScriptShare, SenderBase, SMARTnet, Spectrum Expert, StackWise, The Fastest Way to Increase Your Internet Quotient, TransPath, WebEx, and the WebEx logo are registered trademarks of Cisco Systems, Inc. and/ or its affiliates in the United States and certain other countries.

All other trademarks mentioned in this document or website are the property of their respective owners. The use of the word partner does not imply a partnership relationship between Cisco and any other company. (0903R)

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Contents

Chapter 1: Product Overview and Deployment Guidelines 5

WRP400 Features and Benefits 5

Deployment Models 6

Deploying the WRP400 in a Basic Network 7

Deploying the WRP400 with a Wireless Guest Network 8

Deploying the WRP400 with Mobile Broadband 9

Local Area Network Guidelines 11

Power, Cabling and Telephone Lines 11

Basic Services and Equipment 11

Special Requirements for Voice Deployments 12

Bandwidth for Voice Deployments 12

NAT Mapping for Voice over IP Deployments 14

Local Area Network Design for Voice Deployments 14

WRP400 Maintenance Operations 15

Remote Provisioning 17

Upgrade URL 17

Resync URL 18

Reboot URL 19

Configuration Profile 19

Chapter 2: Configuring Your System for ITSP Interoperability 21

Configuring NAT Mapping 21

Configuring NAT Mapping with a Static IP Address 21

Configuring NAT Mapping with STUN 23

Determining Whether the Router Uses Symmetric or Asymmetric NAT 25

Firewalls and SIP 26

Configuring SIP Timer Values 27

Chapter 3: Configuring Voice Services 28

Understanding Analog Telephone Adapter Operations 28

ATA Software Features 29

Cisco Small Business WRP400 ATA Administration Guide i

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Supported Codecs 29

SIP Proxy Redundancy 30

Other ATA Software Features 31

Contents

Registering to the Service Provider 35

Managing Caller ID Service 37

Optimizing Fax Completion Rates 39

Fax Troubleshooting 40

Silence Suppression and Comfort Noise Generation 41

Configuring Dial Plans 42

About Dial Plans 42

Editing Dial Plans 50

Secure Call Implementation 52

Enabling Secure Calls 52

Secure Call Details 53

Using a Mini-Certificate 54

Generating a Mini Certificate 55

Appendix A: Advanced Voice Fields 57

Info page 57

System page 61

SIP page 62

Regional page 72

Line page 92

User page 111

Appendix B: Data Fields 117

Setup 117

Setup > Basic Setup 118

Setup > DDNS 125

Setup > MAC Address Clone 126

Setup > Advanced Routing 126

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Setup > Mobile Network 127

Setup > Connection Recovery 129

Contents

Wireless Configuration 130

Wireless > Basic Wireless Settings 131

Wireless > Wireless Security 132

Wireless > Wireless MAC Filter 133

Wireless > Advanced Wireless Settings 134

Security 135

Security > Firewall 136

Security > VPN Passthrough 137

Access Restrictions 138

Access Restrictions > Internet Access 138

Applications and Gaming 139

Applications and Gaming > Single Port Forwarding 139

Applications and Gaming > Port Range Forwarding 139

Applications & Gaming > Port Range Triggering 141

Applications & Gaming > DMZ 141

Applications and Gaming > QoS (Quality of Service) 141

Administration 143

Administration > Management 143

Administration > Log 146

Administration > Diagnostics 147

Administration > Factory Defaults 147

Status 148

Status > Router 148

Status > Mobile Network 149

Status > Local Network 150

Status > Wireless Network 150

Appendix C: WRP400 Provisioning Reference 151

Appendix D: Troubleshooting 165

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Contents

Appendix E: Environmental Specifications for the WRP400 169

Appendix F: Where to Go From Here 170

Cisco Small Business WRP400 ATA Administration Guide iv

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Product Overview and Deployment Guidelines

This chapter describes the features and benefits of the WRP400, describes deployment scenarios, and offers guidelines to help you plan your network.

• “WRP400 Features and Benefits,” on page 5

• “Deployment Models,” on page 6

• “Local Area Network Guidelines,” on page 11

• “Special Requirements for Voice Deployments,” on page 12

• “WRP400 Maintenance Operations,” on page 15

• “Remote Provisioning,” on page 17

WRP400 Features and Benefits

With a variety of features, the WRP400 offers the benefits of five devices in one:

1. Router: The WRP400 is a broadband router with a robust security firewall to protect your network.

2. Switch: The WRP400 includes a built-in, 4-port, full-duplex, 10/100 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.

3. Analog Telephone Adapter: The WRP400 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.

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Product Overview and Deployment Guidelines

Deployment Models

4. Wireless Access Point: The WRP400 has an integrated 802.11b/g wireless access point that secures your communications with WEP and WPA security protocols. It is preconfigured to support two wireless networks: one for private use by your business and one for guest use by customers, temporary employees, and other visitors.

5. Mobile Broadband Router: When you attach a compatible Mobile Broadband Modem to the USB port, the WRP400 allows multiple Wi-Fi 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:

www.cisco.com/en/US/products/ps10028/index.html

NOTE Because this device has many unique functions, the administrative tasks for the

WRP400 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 WRP400.

Deployment Models

The versatility of the WRP400 makes it useful for a variety of deployments. Three are described in this section.

• Deploying the WRP400 in a Basic Network, page 7

• Deploying the WRP400 with a Wireless Guest Network, page 8

• Deploying the WRP400 with Mobile Broadband, page 9

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Product Overview and Deployment Guidelines

Private Network

194231

Deployment Models

Deploying the WRP400 in a Basic Network

Internet

Analog phone

Fax

WRP400

Laptop

computer

Printer

Personal

computer

In this scenario, the WRP400 is deployed in a small business that has a basic network configuration.

• The WRP400 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 WRP400 may be configured as an edge device or can be

connected to another device that provides access to the Service Provider network.

• The WRP400 connects the 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 WRP400 phone

ports and have access to the configured Voice over IP services.

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Product Overview and Deployment Guidelines

Internet Access

Device

Wireless Guest Network

Personal

computer

WRP400

Laptop

computer

Analog phone

Fax

Printer

Private Network

Internet

194232

Deployment Models

Deploying the WRP400 with a Wireless Guest Network

In this example, the WRP400 is deployed in an Internet cafe.

• The WRP400 is connected to a cable modem that provides Internet access.

NOTE The WRP400 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 WRP400 by an

Ethernet cable. The manager also has a laptop computer that can be used wirelessly from anywhere on the premises, using the main wireless

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Product Overview and Deployment Guidelines

Mobile Office Network

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*with compatible 3G USB Modem

WRP400*

Wi-Fi Phone

Deployment Models

network, SSID1. The manager and employees using SSID1 have access to the printer. If desired, a wireless phone also could be connected to this network for business use.

• An analog phone and a fax machine are in the private network. The WRP400

is configured for Internet telephone service and for traditional telephone service through a connected phone line.

• The WRP400 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, the customers have no access to the manager’s computer, the printer, or the telephone service.

Deploying the WRP400 with Mobile Broadband

When a compatible mobile broadband modem is connected to the USB port, the WRP400 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 two scenarios illustrated below.

Mobile Office Using the Mobile Network for Internet Access

Laptop

computer

Mobile

network

WRP400

Printer

Wireless Phone

In this example, a team has set up a temporary network at a construction site. The 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).

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Product Overview and Deployment Guidelines

Personal

computer

WRP400

Laptop

computer

Analog phone

Fax

Printer

Private Network

Internet

194233

Mobile

network

Failov

*with compatible 3G USB Modem

WRP400*

Deployment Models

Basic Office Deployment Using the Mobile Network as a Backup Connection

In this example, the business has the same network as illustrated in Deploying the

WRP400 in a Basic Network, page 7. 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 WRP400 fails over to the configured mobile network. When the Internet connection becomes available, the WRP400 recovers the connection.

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Product Overview and Deployment Guidelines

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 WRP400 is intended for deployment

in a very small business. The router is designed to handle the data, voice, and video traffic that would be 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 there is an AC outlet available for every network device that

requires AC power.

- The WRP400 requires power, and Ethernet switches (optional) require

power.

- Some analog telephones require AC power.

• Ethernet cabling: If an Internet access device is present, you will need to

connect it to the WRP400 with an Ethernet cable. You also will need Ethernet cable for any devices that do not have wireless connectivity. It is recommended that Ethernet cables are UTP Cat5e or better.

• PSTN lines: Ensure that the lines are operative and that any features, such as

caller identification, operate properly before starting the installation.

• 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, WRP400, 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

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Product Overview and Deployment Guidelines

Special Requirements for Voice Deployments

• Internet Telephony Service Provider (ITSP) for Voice Over IP telephone service,

supporting a “bring your own device” model

• A computer with Microsoft Windows XP or Windows Vista 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,” on page 12

• “NAT Mapping for Voice over IP Deployments,” on page 14

• “Local Area Network Design for Voice Deployments,” on page 14

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 there is not a Service Level Agreement with regard to the broadband connection quality of service, 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 the utilization levels below 60%, thus reducing 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.

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Product Overview and Deployment Guidelines

Special Requirements for Voice Deployments

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.

For asymmetric connections, such as ADSL, the maximum number of calls is determined by the upstream bandwidth. In general it is a good practice to use no more than 75% of the total available bandwidth for calls. This provides space for data traffic and helps ensure good voice quality.

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 each SIP trunk supports.

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

Codec Approximate bandwidth

budget for each side of conversation

G.711 110 kbps 220

G.726-4087 kbps 174

G.726-3279 kbps 158

G.726-2471 kbps 142

G.726-1663 kbps 126

G.729 55 kbps 110

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

www.erlang.com/calculator/lipb/ www.bandcalc.com/

2 calls 4 calls 6 calls 8 calls

kbps

440 kbps

348 kbps

316 kbps

284 kbps

252 kbps

220 kbps

660 kbps

522 kbps

474 kbps

426 kbps

378 kbps

330 kbps

880 kbps

696 kbps

632 kbps

568 kbps

504 kbps

440 kbps

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Product Overview and Deployment Guidelines

Special Requirements for Voice Deployments

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 three options for providing NAT traversal on your WRP400:

• 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 is providing the edge device, check with your provider to determine if 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 WRP400 is the edge device, the second requirement is met. For more information about the EXT IP setting, see NAT Support Parameters section, page 70.

• 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 running STUN server software, and (3) an edge device with an asymmetric NAT mechanism. If the WRP400 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 NAT Support Parameters

section, page 70.

Local Area Network Design for Voice Deployments

Use the following guidelines to manage the LAN setup for voice deployments.

• Ensure that all telephones are located in the same local area network

subnet.

• Configure your WRP400 as a DHCP server for the purpose of easily adding

network devices to the system. Ensure that the DHCP server can assign

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Product Overview and Deployment Guidelines

WRP400 Maintenance Operations

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 directly connect more than four network devices (other than

wireless devices), you will need to connect an Ethernet switch to the WRP400. 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. The

following settings are recommended:

- Enable Port Fast and Spanning Tree Protocol on the ports to which your

voice devices are connected. The 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. Enabling Port Fast means that the network will be available to the phones when needed. If the switch does not provide a way to enable Port Fast, then you must disable Spanning Tree Protocol.

- In the administrative web pages for the switch, you should enable QoS

and choose DSCP as the Trust Mode.

WRP400 Maintenance Operations

Due to its unique functions, the WRP400 has unique maintenance operations as compared to other Cisco Small Business IP telephony devices.

NOTE For complete instructions about the settings mentioned below, see the WRP400

User Guide.

• Remote Management: For security purposes, remote management is

disabled by default.

- When you first configure the WRP400, connect your administrative

Cisco Small Business WRP400 Administration Guide 15

computer directly to one of the LAN ports and enter the default static IP

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Product Overview and Deployment Guidelines

WRP400 Maintenance Operations

address into your web browser to log on to the configuration utility.

NOTE The default LAN IP address of the WRP400 is 192.168.15.1. If another

device on the network has the same IP address, the WRP400 will take the address 192.168.16.1. You can modify the Local IP Address on the Setup tab > Basic Setup page, Network Setup section.

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

- If you wish to enable web access and wireless access to the

configuration utility, you can use the Administration tab > Management page, Web Access section.

• DHCP Server: The DCHP server is disabled by default. If there are no other

DHCP servers on your network, you can enable the DHCP server option to allow your WRP400 to assign IP addresses to connected devices automatically. This setting is on the Setup tab > Basic Setup page, DHCP Server Setting section.

• System Logging: If you wish to enable system logging, be aware that there

are two sets of system logs: one for the data (router) functions and another for the voice functions.

- Data (router) logging: See the Administration tab > Logging page.

- Voice logging: See the Voice tab > System page, Miscellaneous

Settings section.

• Factory Reset: If you wish to reset your WRP400 to the factory default

settings, you can 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, and then click

Administration > Factory Defaults. Next to Restore Router Factory Defaults, click Yes . Then click Save Settings to begin the operation.

- Option 2: Press and hold the reset button located on the side panel for

Cisco Small Business WRP400 Administration Guide 16

approximately ten seconds.

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Product Overview and Deployment Guidelines

Remote Provisioning

Factory reset of voice settings: Use one of the following methods:

- Option 1: Log on to the configuration utility, and then click

Administration tab > Factory Defaults. Next to Restore Voice Factory Defaults, click Yes . Then click Save Settings 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 and then press 1 to confirm or * to cancel. After you hear “Option successful,” you can hang up the phone.

Remote Provisioning

Like other Cisco Small Business IP Telephony Devices, the WRP400 provides for secure provisioning and remote upgrade. Provisioning is achieved through configuration profiles transferred to the device via TFTP, HTTP, or HTTPS. To configure Provisioning, go to the Provisioning tab in the Configuration Utility.

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

www.cisco.com/en/US/docs/voice_ip_comm/csbpvga/ata/provisioning/guide/ Cisco_Small_Business_IP_Telephony_Provisioning_Guide.pdf

Upgrade URL

Remote firmware upgrade is achieved via TFTP or HTTP (firmware upgrades using HTTPS are not supported). Remote upgrades are initiated by causing the WRP400 to request the upgrade firmware image by providing a URL for the WRP400 to retrieve the firmware.

NOTE If the value of the

cannot upgrade the WRP400 even if the web page indicates otherwise.

The syntax of the Upgrade URL is as follows:

http://WRP400_ip_address/admin/upgrade?[protocol://][server- name[:port]][/firmware-pathname]

Upgrade Enable

parameter in the Provisioning page is No, you

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Product Overview and Deployment Guidelines

Remote Provisioning

Both HTTP and TFTP are supported for the upgrade operation.

protocol

If no host that requests the URL is used as

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

The

firmware-pathname

directory on the TFTP or HTTP server. If no

spa.bin

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

is specified, TFTP is assumed. If no

server-name

is typically the file name of the binary located in a

firmware-pathname

is assumed, as in the following example:

server-name

is specified, the

is specified,

Resync URL

The WRP400 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.

The Resync URL lets you force the WRP400 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://WRP400_ip_address/admin/resync?[[protocol://][server- name[:port]]/profile-pathname]

NOTE The WRP400 resyncs only when it is idle.

If no parameter follows page is used.

protocol

If no host that requests the URL is used as

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.cfg

is specified, TFTP is assumed. If no

/resync?,

the Profile Rule setting from the Provisioning

server-name

is specified, the

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Product Overview and Deployment Guidelines

Remote Provisioning

Reboot URL

The Reboot URL lets you reboot the WRP400. The Reboot URL is as follows:

http://WRP400_ip_address/admin/reboot

NOTE The WRP400 reboots only when it is idle.

Configuration Profile

Because the WRP400 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. Binary files are not supported for the configuration of data (router) parameters. For more information about the data parameters, see Appendix B, “Data Fields.”

• Voice parameters: Use the binary or 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 WRP400. For more information about the data parameters, see Appendix A,

“Advanced Voice Fields.”

NOTE You can download the SPC at the following URL: tools.cisco.com/

support/ downloads/go/Redirect.x?mdfid=282414113

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Product Overview and Deployment Guidelines

Remote Provisioning

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> <Admin_Passwd>some secret</Admin_Passwd> <Upgrade_Enable>Yes</Upgrade_Enable> </flat-profile>

The names of parameters in XML profiles can generally be inferred from the WRP400 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. For more information, see Appendix C, “WRP400 Provisioning Reference.”

Binary Format

Binary format profiles contain voice parameter values and user access permissions for the parameters. By convention, the profile uses the extension .cfg (for example, spa2102.cfg). The Profile Compiler (SPC) tool compiles a plain-text file containing parameter-value pairs into a properly formatted and encrypted .cfg file.

The syntax of the plain-text file accepted by the profile compiler is a series of parameter-value pairs, with the value in double quotes. Each parameter-value pair is followed by a semicolon. Here is an example of a valid text source profile for input to the SPC tool:

Admin_Passwd “some secret”; Upgrade_Enable “Yes”;

The names of parameters in the source text files for the SPC tool can generally be inferred from the WRP400 Configuration Utility, by substituting underscores (_) for spaces and other control characters. To distinguish between Line 1, 2, 3, and 4, corresponding parameter names are augmented by adding [1], [2], [3], or [4]. For example, the Line 1 Proxy is named Proxy[1] in source text profiles for input to the SPC.

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Configuring Your System for ITSP Interoperability

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

• “Configuring NAT Mapping,” on page 21

• “Firewalls and SIP,” on page 26

• “Configuring SIP Timer Values,” on page 27

Configuring NAT Mapping

As discussed in Chapter 1, “Product Overview and Deployment Guidelines,” some form of NAT mapping is needed to support VoIP. If your ITSP does not support NAT mapping through a Session Border Controller, and your edge device is not a SIP-ALG router, you can address this issue through one of the following methods:

• “Configuring NAT Mapping with a Static IP Address,” on page 21

• “Configuring NAT Mapping with STUN,” on page 23

Configuring 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 WRP400 is the edge device, this requirement is met. If another device is used as the edge device, see “Determining Whether the Router Uses Symmetric or

Asymmetric NAT,” on page 25.

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Configuring Your System for ITSP Interoperability

Configuring NAT Mapping

• If the WRP400 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 WRP400 is connected.

NOTE Use NAT mapping only if the ITSP network does not provide a Session Border

Controller functionality.

STEP 1 Start Internet Explorer, connect to the Configuration Utility, and choose Voice >

Admin Login. 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 SIP.

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

• Handle VIA received, Insert VIA received, Substitute VIA Addr: Choose yes.

• Handle VIA rport, Insert VIA rport, Send Resp To Src Port: Choose yes.

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

Voice tab > SIP: NAT Support Parameters

STEP 4

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

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

• NAT Mapping Enable: Choose yes.

• NAT Keep Alive Enable: Choose yes.

Cisco Small Business WRP400 Administration Guide 22

Voice tab > Line N > NAT Settings

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Configuring Your System for ITSP Interoperability

Configuring NAT Mapping

STEP 6 Click Save Settings.

NOTE You also need to configure the firewall settings on your router to allow SIP

traffic. See “Firewalls and SIP,” on page 26.

Configuring 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 WRP400 is the

edge device, this requirement is not met. For more information, see

“Determining Whether the Router Uses Symmetric or Asymmetric NAT,” on page 25.

• If the WRP400 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 WRP400 is connected.

NOTE Use NAT mapping only if the ITSP network does not provide a Session Border

Controller functionality.

STEP 1 Start Internet Explorer, connect to the Configuration Utility, choose Voice > Admin

Login. 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 SIP.

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

• Handle VIA received: yes

• Handle VIA rport: yes

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Configuring Your System for ITSP Interoperability

Configuring NAT Mapping

• Insert VIA received: yes

• Insert VIA rport: yes

• Substitute VIA Addr: yes

• Send Resp To Src Port: yes

• STUN Enable: Choose yes.

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

Voice tab > SIP > NAT Support Parameters

STEP 4

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

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

• NAT Mapping Enable: Choose yes.

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

Voice tab > Line N > NAT Settings

NOTE Your ITSP may require the WRP400 to send NAT keep alive messages to

keep the NAT ports open permanently. Check with your ITSP to determine the requirements.

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Configuring Your System for ITSP Interoperability

Configuring NAT Mapping

STEP 6 Click Save Settings.

NOTE You also need to configure the firewall settings on your router to allow SIP

traffic. See “Firewalls and SIP,” on page 26.

Determining 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, then 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 you do not have firewall running on your computer that could block the

syslog port (port 514 by default).

STEP 2 Start Internet Explorer, connect to the Configuration Utility, choose Voice > Admin

Login. If prompted, enter the administrative login provided by the Service Provider. (The default username and password are both admin.)

STEP 3 To enable debugging, complete the following tasks:

a. Under the Voice menu, click System.

b. In the Debug Server field, enter the IP address of your syslog server. This

address and port number must be reachable from the WRP400.

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Configuring Your System for ITSP Interoperability

Firewalls and SIP

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

STEP 4 To collect information about the type of NAT 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 signalling, complete the following task:

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

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

STEP 6 Click Save Settings.

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.

you want to modify.

• Make sure that the following ports are not blocked:

• SIP ports—UDP port 5060 through 5063, which are used for the ITSP line

interfaces

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Configuring Your System for ITSP Interoperability

Configuring SIP Timer Values

• RTP ports—16384 to 16482

• Also disable SPI (Stateful Packet Inspection) if this function exists on your

firewall.

Configuring 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 ISTP. 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 ”SIP Timer Values (sec) section,” on page 65 of

Appendix A.

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Configuring Voice Services

This chapter describes how to configure your WRP400 to meet the customer’s requirements for voice services.

• “Understanding Analog Telephone Adapter Operations,” on page 28

• “Managing Caller ID Service,” on page 37

• “Silence Suppression and Comfort Noise Generation,” on page 41

• “Configuring Dial Plans,” on page 42

• “Secure Call Implementation,” on page 52

Understanding Analog Telephone Adapter Operations

The WRP400 is equipped with a built-in Analog Telephone Adapter (ATA). An ATA is an intelligent low-density Voice over IP (VoIP) gateway that enables carrierclass residential and business IP Telephony services delivered over broadband or high-speed Internet connections. Users can access Internet phone services using standard analog telephone equipment. In addition, the WRP400 has two line ports that can be connected to the Public Switched Telephone Network (PSTN) so that your business can support legacy phone numbers and fax numbers.

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Configuring Voice Services

252075

ATA S of t w ar e F ea tu r es

The WRP400 maintains the state of each call it terminates and makes the proper reaction to user input events (such as on/off hook or hook flash). The WRP400 uses the Session Initiation Protocol (SIP) open standard, so there is little or no involvement by a “middle-man” server or media gateway controller. SIP allows interoperation with all ITSPs that support SIP.

elephone/fax

WRP400

Ethernet

Internet

Access Device

Internet

Service Provider

VoIP Infrastructure

SIP proxy

Voice

gateway

PSTN

Phone

ATA Software Features

The WRP400 is equipped with a full featured, fully programmable ATA that can be custom provisioned within a wide range of configuration parameters. The following sections describe the factors that contribute to voice quality:

• “Supported Codecs,” on page 29

• “SIP Proxy Redundancy,” on page 30

• “Other ATA Software Features,” on page 31

Supported Codecs

The WRP400 supports the following codecs:

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

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

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ATA S of t w ar e F ea tu r es

• G.726-32

• G.729a

The administrator can select the preferred codecs to be used for each line. See

“Audio Configuration section,” on page 104.

In addition, negotiation of the optimal voice codec sometimes depends on the ability of an ATA to match a codec name with the codec used by the far-end device. You can individually name the various codecs so that the WRP400 can successfully negotiate the codec with the far-end equipment. For more information, see Audio Configuration section, page 104.

This low complexity codec supports compressed 16, 24, 32, and 40 kbps digitized voice transmission at one through ten 10 ms voice frames per packet. This codec provides high voice quality.

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.

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 WRP400 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 WRP400 can be instructed to contact a SIP proxy server in a domain named in the SIP message. The WRP400 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 WRP400 tries to contact the list of hosts in the order of their stated priority.

If the WRP400 is currently using a lower priority proxy server, it periodically probes the higher priority proxy to see 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. See

Appendix B, “Data Fields.”.

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ATA S of t w ar e F ea tu r es

Other ATA Software Features

The following table summarizes other features provided by the WRP400.

Feature Description

Silence Suppression

Modem and Fax Pass-Through

Adaptive Jitter Buffer

See “Silence Suppression and Comfort Noise

Generation,” on page 41.

Modem pass-through mode can be triggered only by

•

predialing the number set in the (Set in the Regional tab.)

• FAX pass-through mode is triggered by a CED/CNG tone or

an NSE event.

• Echo canceller is automatically disabled for Modem pass-

through mode.

• Echo canceller is disabled for FAX pass-through if the

parameter for that line (in that case FAX pass-through is the same as Modem pass-through).

• Call waiting and silence suppression is automatically

disabled for both FAX and Modem pass-through. In addition, out-of-band DTMF Tx is disabled during modem or fax passthrough.

The WRP400 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.

FAX Disa bl e E CA N

Modem Line Toggle Code.

(Line 1 or 2 tab) is set to “yes”

International Caller ID Delivery

Cisco Small Business WRP400 Administration Guide 31

The WRP400 has a Network Jitter Level control setting for each line of service. The jitter level determines how aggressively the WRP400 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 “Advanced Voice Fields,” on page 57.

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

“Advanced Voice Fields,” on page 57.

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ATA S of t w ar e F ea tu r es

Feature Description

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 “Secure Call Implementation” section

on page 52.

Adjustable Audio Frames Per Packet

DTMF The WRP400 may relay DTMF digits as out-of-band events

Call Progress Tone Generation

This feature allows the user to set the number of audio frames contained in one RTP packet. Packets can be adjusted to contain from 1–10 audio frames. Increasing the number of packets decreases the bandwidth utilized, but it also increases delay and may affect voice quality. See the RTP Packet Size parameter found in the SIP tab in the

“Advanced Voice Fields,” on page 57.

to preserve the fidelity of the digits. This can enhance the reliability of DTMF transmission required by many IVR applications such as dial-up banking and airline information. DTMF is configured in the parameter found in the Line tabs. See the “Advanced

Voice Fields,” on page 57.

The WRP400 has configurable call progress tones. Call progress tones are generated locally on the WRP400 so an end user is advised of status (such as ringback). Parameters for each type of tone (for instance a dial tone played back to an end user) may include frequency and amplitude of each component, and cadence information. See the Regional tab in the “Advanced Voice Fields,” on

page 57.

DTMF Tx Mode

Call Progress Tone Pass Through

Echo Cancellation Impedance mismatch between the telephone and the IP

Cisco Small Business WRP400 Administration Guide 32

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 the “Adv anc ed Vo ice

Fields,” on page 57.

Telephony gateway phone port can lead to near-end echo. The WRP400 has a near-end echo canceller that compensates for impedance match. The WRP400 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 “Advanced Voice

Fields,” on page 57.

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Configuring Voice Services

ATA S of t w ar e F ea tu r es

Feature Description

Signaling Hook Flash Event

Configurable Dial Plan with Interdigit Time rs

The WRP400 can signal hook flash events to the remote party on a connected call. This feature can be used to provide advanced mid-call services with third-party-callcontrol. 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 hookflash event to the softswitch:

•

Call Waiting Service (parameter Line tab)

• Three Way Conference Service (parameter

set in the Line tab)

serv

• Three Way Call Service (parameter

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

“Advanced Voice Fields,” on page 57.

The WRP400 has three configurable interdigit timers:

Initial timeout (T)—Signals that the handset is off the hook

•

and that no digit has been pressed yet.

call waiting serv

three-way conf

three-way call serv

set in the

set

• 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 “Configuring Dial Plans,” on page 42 for more information.

Polarity Control The WRP400 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 “Advanced Voice Fields,”

on page 57.

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Configuring Voice Services

ATA S of t w ar e F ea tu r es

Feature Description

Calling Party Control

Report Generation and Event Logging

Syslog and Debug Server Records

Calling Party Control (CPC) signals to the called party equipment that the calling party has hung up during a connected call by removing the voltage between the tip and ring momentarily. This feature is useful for autoanswer equipment, which then knows when to disengage. CPC is configured in the Regional, Line, and PSTN Line tabs. See “Advanced Voice Fields,” on page 57.

The WRP400 reports a variety of status and error reports to assist service providers to diagnose problems and evaluate the performance of their services. The information can be queried by an authorized agent, using HTTP with digested authentication, for instance. The information may be organized as an XML page or HTML page. Report Generation and Event Logging are configured in the System, Line, and PSTN Line tabs. See

“Advanced Voice Fields,” on page 57.

Syslog and Debug Sever Records log more details than Report Generation and Event Logging. Using the configuration parameters, the WRP400 allows you to select which type of activity/events should be logged. Syslog and Debug Server allow the information captured to be sent to a Syslog Server. Syslog and Debug Server Records are configured in the System, Line, and PSTN Line tabs. See “Advanced Voice Fields,” on page 57.

SIP Over TLS The WRP400 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 “Advanced Voice Fields,” on

page 57.

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Configuring Voice Services

Registering to the Service Provider

To use VoIP phone service, you must configure your WRP400 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 Start Internet Explorer, connect to the Configuration Utility, choose Voice > Admin

Provider. (The default username and password are both admin.)rovided by your Service Provider.

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 Save Settings. The devices reboot.

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Configuring Voice Services

Registering to the Service Provider

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

• Under the Voice menu, click Info. Scroll down to the

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.

Line 1 Status or Line 2

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Configuring Voice Services

Managing Caller ID Service

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

Parameter Ta b 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.

Caller ID FSK Standard

Cisco Small Business WRP400 Administration Guide 37

Regional

• DTMF (Denmark) With PR—CID only. DTMF sent after

polarity reversal (and no DTAS) and before first ring.

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

The WRP400 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.

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Configuring Voice Services

Polarity

Reversal

First

Ring

CAS

(DTAS)

DTMF/

FSK

Polarity

Reversal

CAS

(DTAS)

FSK

CAS

(DTAS)

Wait For

ACK

FSK

First

Ring

FSK

OSI FSK

a) Bellcore/ETSI Onhook Post-Ring FSK

d) Bellcore Onhook FSK w/o Ring

f) Bellcore/ETSI Offhook FSK

c) ETSI Onhook Pre-Ring FSK/DTMF

e) ETSI Onhook FSK w/o Ring

DTMF

b) ETSI Onhook Post-Ring DTMF

First Ring

Managing Caller ID Service

There are three types of Caller ID:

• On Hook Caller ID Associated with Ringing — This type of Caller ID is used

for incoming calls when the attached phone is on hook. See the following figure (a) – (c). All CID methods can be applied for this type of CID.

• On Hook Caller ID Not Associated with Ringing — This feature is used to

send VMWI signal to the phone to turn the message waiting light on and off (see Figure 1 (d) and (e)). 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).

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Configuring Voice Services

Optimizing Fax Completion Rates

Issues can occur with fax transmissions over IP networks, even with the T.38 standard, which is supported by the WRP400. You can adjust several settings on your WRP400 to optimize your fax completion rates.

NOTE Only T.38 Fax is supported. The WRP400 supports one connection.

STEP 1 Ensure that you have enough bandwidth for the uplink and the downlink.

• For G.711 fallback, it is recommend to have approximately 100Kbps.

• 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:

• Network Jitter Buffer: very high

• Jitter buffer adjustment: disable

• Call Waiting: no

• 3 Way Calling: no

• Echo Canceller: no

• Silence suppression: no

• Preferred Codec: G.711

• Use pref. codec only: yes

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 WRP400 by configuring the following parameter on the Line

tab for the FXS port to which the FAX machine is connected:

FAX_Passthru_Method: ReINVITE

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Configuring Voice Services

Optimizing Fax Completion Rates

NOTE If a T.38 call cannot be set-up, then the call automatically reverts to G.711

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

fallback.

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 D,

“Troubleshooting.”

NOTE You may also capture data using a sniffer trace.

STEP 7 Identify the type of fax machine connected to the ATA device.

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Configuring Voice Services

Silence Suppression and Comfort Noise Generation

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 supported by the network by reducing the required bandwidth for a single call. VAD uses a sophisticated algorithm to distinguish between speech and non-speech signals. Based on the current and past statistics, the VAD algorithm decides whether or not speech is present. If the VAD algorithm decides speech is not present, the silence suppression and comfort noise generation is activated. This is accomplished by removing and not transmitting the natural silence that occurs in 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 created by the VAD and Silence Suppression feature.

Silence suppression is configured in the Line and PSTN Line tabs. See

Appendix B, “Data Fields.”

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Configuring Dial Plans

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

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

• “About Dial Plans,” on page 42

• “Editing Dial Plans,” on page 50

About Dial Plans

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

Refer to the following topics:

• “Digit Sequences,” on page 42

• “Digit Sequence Examples,” on page 44

• “Acceptance and Transmission the Dialed Digits,” on page 47

• “Dial Plan Timer (Off-Hook Timer),” on page 48

• “Interdigit Long Timer (Incomplete Entry Timer),” on page 49

• “Interdigit Short Timer (Complete Entry Timer),” on page 49

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.

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Configuring Dial Plans

Digit Sequence Function

0 1 2 3 4 5 6 7 8 9 0 * #

x Enter x to represent any character on the phone

[sequence] Enter characters within square brackets to create

(period)

<dialed:substituted> Use this format to indicate that certain dialed

Enter any of these characters to represent a key that the user must press on the phone keypad.

keypad.

a list of accepted key presses. The user can press any one of the keys in the list.

Numeric range

•

For example, you would enter user to press any one digit from 2 through 9.

• Numeric range with other characters

For example, you would enter the user to press 3, 5, 6, 7, 8, or *.

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.

digits are replaced by other characters when the sequence is transmitted. The dialed digits can be zero or more characters.

[2-9] to allow the

[35-8*] to allow

Cisco Small Business WRP400 Administration Guide 43

EXAMPLE 1: <8:1650>xxxxxxx

When the user presses 8 followed by a sevendigit 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

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Configuring Dial Plans

Digit Sequence Function

(comma)

(exclamation point)

*xx

S0 or L0

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.

Enter an exclamation point to prohibit a dial sequence pattern.

EXAMPLE: 1900xxxxxxx!

The system rejects any 11-digit sequence that begins with 1900.

Enter an asterisk to allow the user to enter a 2digit star code.

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.

• Extensions on your system

[1-8]xx Allows a user dial any three-digit number that starts with the digits 1

through 8. If your system uses four-digit extensions, you would instead enter the following string:

[1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8,

[1-8]xx x

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• Local dialing with seven-digit number

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

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 10digit 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 |

<:1212>xxxxxxx

011xxxxxx. | 0 | [49]11 )

| 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9,

9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8,

9, <:1>[2-9]xxxxxxxxx | 8,

8, <:1212>xxxxxxx This is example is useful where a local area code is required

by the carrier but the majority of calls go to one area code. After the user presses 8, an external dial tone sounds. The user can enter any seven-digit number. The system automatically inserts the 1 prefix and the 212 area code before transmitting the number to the carrier.

• U.S. long distance dialing

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.

9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxx xx ! | 9,

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• Blocked number

9, 1 900 xxxxxxx ! This digit sequence is useful if you want to prevent users from

dialing numbers that are associated with high tolls or inappropriate content, such as 1-900 numbers in the U.S.. After the user press 9, an external dial tone sounds. If the user enters an 11-digit number that starts with the digits 1900, the call is rejected.

• U.S. international dialing

9, 011xxxxxx. | 0 | [49]11 )

! |

9, 01 1xxxxxx. 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.

9, 1 900 xxxxxxx ! |

• Informational numbers

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.

0 | [49]11 )

9, <:1>[2-9]xxxx xxxxx | 8,

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Acceptance and Transmission the Dialed Digits

When a user dials a series of digits, each sequence in the dial plan is tested as a possible match. The matching sequences form a set of candidate digit sequences. As more digits are entered by the user, the set of candidates diminishes until only one or none are valid. When a terminating event occurs, the WRP400 either accepts the user-dialed sequence and initiates a call, or else rejects the sequence as invalid. The user hears the reorder (fast busy) tone if the dialed sequence is invalid.

The following table explains how terminating events are processed.

Ter min ati ng E ven t Processing

The dialed digits do not match any sequence in the dial plan.

The dialed digits exactly match one sequence in the dial plan.

A timeout occurs. The number is rejected if the dialed digits are

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

The number is rejected.

•

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.

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.

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.

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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: (Ps<: n> | dial plan )

• s: The number of seconds; if no number is entered after P, the default timer of 5

seconds applies.

• n: (optional): The number to transmit automatically when the timer expires; you

can enter 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>, then 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: ( | 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: ( 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:

P9 | (9,8<:1408>[2-9]xxxxx x | 9,8,1[2-9]xxxxxxxxx

P9<:23> | (9,8<:1408>[2-9]xxxxxx | 9,8,1[2-

( 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.

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NOTE This section explains how to edit a timer as part of a dial plan. Alternatively, you can

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.

modify the Control Timer that controls the default interdigit timers for all calls. See

“Resetting the Control Timers,” on page 51.

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.

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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:

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 | 9,8,011xx. | 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.

S:6, (9,8<:1408>[2-9]xxxx xx | 9,8,1[2-9]xxxxxxxxx |

9,8,1[2-9]xxxxxxxxxS0 |

Editing Dial Plans

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

Entering the Line Interface Dial Plan

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

STEP 1 Start Internet Explorer, connect to the Configuration Utility, choose Voice > Admin

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STEP 2 Under the Voice menu, click Line 1 or Line 2, depending on the line interface that

STEP 3 Scroll down to the Dial Plan section.

STEP 4 Enter the digit sequences in the Dial Plan field. For more information, see “A b ou t

STEP 5 Click Save Settings.

Provider. (The default username and password are both admin.)rovided by your Service Provider.

you want to configure.

Dial Plans,” on page 42.

Resetting 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 “About Dial Plans,” on page 42.

STEP 1 Start Internet Explorer, connect to the Configuration Utility, choose Voice > Admin

Provider. (The default username and password are both admin.)rovided by your Service Provider.

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.

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Secure Call Implementation

This section describes secure call implementation with the WRP400 . It includes the following topics:

• “Enabling Secure Calls” section on page 52

• “Secure Call Details” section on page 53

• “Using a Mini-Certificate” section on page 54

• “Generating a Mini Certificate” section on page 55

NOTE This is an advanced topic meant for experience installers. Also see the

Guide

www.cisco.com/en/US/docs/voice_ip_comm/csbpvga/ata/provisioning/guide/ Cisco_Small_Business_IP_Telephony_Provisioning_Guide.pdf

at the following URL:

Enabling Secure Calls

A secure call is established in two stages. The first stage is no different from normal call setup. The second stage starts after the call is established in the normal way with both sides ready to stream RTP packets.

In the second stage, the two parties exchange information to determine if the current call can switch over to the secure mode. The information is transported by base64 encoding embedded in the message body of SIP INFO requests, and responses using a proprietary format. If the second stage is successful, the WRP400 plays a special Secure Call Indication Tone for a short time to indicate to both parties that the call is secured and that RTP traffic in both directions is being encrypted.

If the user has a phone that supports call waiting caller ID (CIDCW) and that service is enabled, the CID will be updated with the information extracted from the Mini-Certificate received from the remote party. The Name field of the CID will be prepended with a ‘$’ symbol. Both parties can verify the name and number to ensure the identity of the remote party.

Provisioning

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The signing agent is implicit and must be the same for all ATAs that communicate securely with each other. The public key of the signing agent is pre-configured into the WRP400 by the administrator and is used by the WRP400 to verify the MiniCertificate of its peer. The Mini-Certificate is valid if it has not expired, and it has a valid signature.

The WRP400 can be configured so that, by default, all outbound calls are either secure or not secure. If secure by default, the user has the option to disable security when making a call by dialing *19 before dialing the target number. If not secure by default, the user can make a secure outbound call by dialing *18 before dialing the target number. However, the user cannot force inbound calls to be secure or not secure; that depends on whether the caller has security enabled or not.

The WRP400 will not switch to secure mode if the CID of the called party from its Mini-Certificate does not agree with the user-id used in making the outbound call. The WRP400 performs this check after receiving the Mini-Certificate of the called party

Secure Call Details

Looking at the second stage of setting up a secure call in greater detail, this stage can be further divided into two steps.

STEP 1 The caller sends a “Caller Hello” message (base64 encoded and embedded in the

message body of a SIP INFO request) to the called party with the following information:

• Message ID (4B)

• Version and flags (4B)

• SSRC of the encrypted stream (4B)

• Mini-Certificate (252B)

Upon receiving the Caller Hello, the called party responds with a Callee Hello message (base64 encoded and embedded in the message body of a SIP response to the caller’s INFO request) with similar information, if the Caller Hello message is valid. The caller then examines the Callee Hello and proceeds to the next step if the message is valid.

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STEP 2 The caller sends the “Caller Final” message to the called party with the following

information:

• Message ID (4B)

• Encrypted Master Key (16B or 128b)

• Encrypted Master Salt (16B or 128b)

Using a Mini-Certificate

The Master Key and Master Salt are encrypted with the public key from the called party mini-certificate. The Master Key and Master Salt are used by both ends for deriving session keys to encrypt subsequent RTP packets. The called party then responds with a Callee Final message (which is an empty message).

The Mini-Certificate (MC) contains the following information:

• User Name (32B)

• User ID or Phone Number (16B)

• Expiration Date (12B)

• Public Key (512b or 64B)

• Signature (1024b or 512B)

The MC has a 512-bit public key used for establishing secure calls. The administrator must provision each subscriber of the secure call service with an MC and the corresponding 512-bit private key. The MC is signed with a 1024-bit private key of the service provider, which acts as the CA of the MC. The 1024-bit public key of the CA signing the MC must also be provisioned for each subscriber.

The CA public key is used to verify the MC received from the other end. If the MC is invalid, the call will not switch to secure mode. The MC and the 1024-bit CA public key are concatenated and base64 encoded into the single parameter

Certificate

parameter, which should be kept secret, like a password. (

SRTP Private Key

. The 512-bit private key is base64 encoded into the

are configured in the Line tabs.)

SRTP Private Key

Mini Certificate

and

Mini

Because the secure call establishment relies on exchange of information embedded in message bodies of SIP INFO requests/responses, the service provider must ensure that the network infrastructure allows the SIP INFO messages to pass through with the message body unmodified.

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NOTE The partner sites require a logon.

Generating a Mini Certificate

Cisco provides a Mini Certificate Generator for the generation of mini certificates and private keys. Partners can download the Mini Certificate Generator by going to Cisco Partner Central, Voice & Conferencing page, Technical Resources section. Use the following URL:

www.cisco.com/web/partners/sell/smb/products/ voice_and_conferencing.html#~vc_technical_resources

The Mini Certificate Generator uses the following syntax:

gen_mc ca-key user-name user-id expire-date

Where:

• ca-key is a text file with the base64 encoded 1024-bit CA private/public key

pairs for signing/verifying the MC, such as the following:

9CC9aYU1X5lJuU+EBZmi3AmcqE9U1LxEOGwopaGyGOh3VyhKgi6JaVtQZt87PiJINKW8XQj3B9Qq e3VgYxWCQNa335YCnDsenASeBxuMIEaBCYd1l1fVEodJZOGwXwfAde0MhcbD0kj7LVlzcsTyk2TZ YTccnZ75TuTjj13qvYs=5nEtOrkCa84/mEwl3D9tSvVLyliwQ+u/ Hd+C8u5SNk7hsAUZaA9TqH8Iw0J/ IqSrsf6scsmundY5j7Z5mK5J9uBxSB8t8vamFGD0pF4zhNtbrVvIXKI9kmp4vph1C5jzO9gDfs3M F+zjyYrVUFdM+pXtDBxmM+fGUfrpAuXb7/k=

• user-name is the name of the subscriber, such as “Joe Smith”. Maximum length

is 32 characters

• user-id is the User ID of the subscriber, which must match exactly the user-id

used in the INVITE when making the call, such as “14083331234”. The maximum length is 16 characters.

• expire-date is the expiration date of the MC, such as “00:00:00 1/1/34”

(34=2034). Internally the date is encoded as a fixed 12B string: 000000010134

The tool generates the be provisioned.

Mini Certificate

and

SRTP Private Key

parameters that can

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EXAMPLE:

gen_mc ca_key “Joe Smith” 14085551234 “00:00:00 1/1/34”

This example produces the following Mini Certificate and SRTP Private Key:

<Mini Certificate> Sm9lIFNtaXRoAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAxNDA4NTU1MTIzNAAAAAAAMDAwMDAwMDEw MTM00OvJakde2vVMF3Rw4pPXL7lAgIagMpbLSAG2+++YlSqt198Cp9rP/ xMGFfoPmDKGx6JFtkQ5sxLcuwgxpxpxkeXvpZKlYlpsb28L4Rhg5qZA+Gqj1hDFCmG6dffZ9SJhx ES767G0JIS+N8lQBLr0AuemotknSjjjOy8c+1lTCd2t44Mh0vmwNg4fDck2YdmTMBR516xJt4/ uQ/ LJQlni2kwqlm7scDvll5k232EvvvVtCK0AYa4eWd6fQOpiESCO9CC9aYU1X5lJuU+EBZmi3AmcqE 9U1LxEOGwopaGyGOh3VyhKgi6JaVtQZt87PiJINKW8XQj3B9Qqe3VgYxWCQNa335YCnDsenASeBx uMIEaBCYd1l1fVEodJZOGwXwfAde0MhcbD0kj7LVlzcsTyk2TZYTccnZ75TuTjj13qvYs= <SRTP Private Key> b/DWc96X4YQraCnYzl5en1CIUhVQQqrvcr6Qd/8R52IEvJjOw/ e+Klm4XiiFEPaKmU8UbooxKG36SEdKusp0AQ==

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Advanced Voice Fields

This appendix describes the Advanced settings that are available after you login from the Voice > Admin Login page.

NOTE For information about the other pages in the Configuration Utility, see the WRP400

User Guide.

Info page

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

• “Info page,” on page 57

• “System page,” on page 61

• “SIP page,” on page 62

• “Regional page,” on page 72

• “Line page,” on page 92

• “User page,” on page 111

You can use the Voice tab > Info page to view information about the WRP400. This page includes the following sections:

• “Product Information section,” on page 58

• “System Status section,” on page 58

• “Line Status section,” on page 59

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

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Info page

Voice tab > Info page >

Product Information section

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 if the

WRP400 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 tab > Info page >

System Status section

Current Time Current date and time of the system; for example, 10/3/

2003 16:43:00.

Elapsed Time Total time elapsed since the last reboot of the system; for

example, 25 days and 18:12:36.

RTP Packets Sent Total number of RTP packets sent (including redundant

packets).

RTP Bytes Sent Total number of RTP bytes sent.

RTP Packets Recv Total number of RTP packets received (including redundant

packets).

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).

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Info page

Current Time Current date and time of the system; for example, 10/3/

2003 16:43:00.

SIP Bytes Recv Total number of bytes of SIP messages received (including

retransmissions).

External IP External IP address used for NAT mapping.

Voice tab > Info page >

Line Status section

(PSTN) Hook State Hook state of the FXO port. Options are either On or Off.

Registration State Indicates if 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 FXO 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

• Collecting PSTN Pin

• Invalid PSTN PIN

• PSTN Caller Accepted

Call 1 and 2 Tone Type of tone used by the call.

Call 1 and 2 Encoder

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• Connected to PSTN

Codec used for encoding.

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Info page

Call 1 and 2 Decoder

Call 1 and 2 FAX Status of the fax pass-through mode.

Call 1 and 2 Type Direction of the call. May take one of the following values:

Call 1 and 2 Remote Hold

Call 1 and 2 Callback

Codec used for decoding.

PSTN Gateway Call = VoIP-To-PSTN Call

•

• VoIP Gateway Call = PSTN-To-VoIP Call

• PSTN To Line 1 = PSTN call ring through and answered by

Line 1

• Line 1 Forward to PSTN Gateway = VoIP calls Line 1 then

forwarded to PSTN GW

• Line 1 Forward to PSTN Number =VoIP calls Line 1 then

forwarded to PSTN number

• Line 1 To PSTN Gateway

• Line 1 Fallback To PSTN Gateway

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

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

Call 1 and 2 Peer Name

Call 1 and 2 Peer Phone

Call 1 and 2 Call Duration

Call 1 and 2 Packets Sent

Call 1 and 2 Packets Recv

Call 1 and 2 Bytes Sent

Call 1 and 2 Bytes Recv

Call 1 and 2 Decode Latency

Call 1 and 2 Jitter Number of milliseconds for receiver jitter.

Name of the internal phone.

Phone number of the internal phone.

Duration of the call.

Number of packets sent.

Number of packets received.

Number of bytes sent.

Number of bytes received.

Number of milliseconds for decoder latency.

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System page

Call 1 and 2 Round Trip De lay

Call 1 and 2 Packets Lost

Call 1 and 2 Packet Error

Call 1 and 2 Mapped RTP Port

Call 1 and 2 Media Loopback

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

• “System Configuration section” section on page 61

Number of milliseconds for delay.

Number of packets lost.

Number of invalid packets received.

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

Media loopback is used to quantitatively and qualitatively measure the voice quality experienced by the end user.

• “Miscellaneous Settings section” section on page 62

Voice tab > System page >

System Configuration section

Restricted Access Domains

Enable Web Admin Access

Admin Password Password for the administrator. The default is no password.

User Password Password for the user. The default is no password.

This feature is used when implementing software customization.

Lets you enable or disable local access to the Configuration Utility. Select yes or no from the drop-down menu.

The default is yes.

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SIP page

Voice tab > System page >

Miscellaneous Settings section

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.

SIP page

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

• “SIP Parameters section” section on page 63

• “SIP Timer Values (sec) section” section on page 65

• “Response Status Code Handling section” section on page 67

• “RTP Parameters section” section on page 67

• “SDP Payload Types section” section on page 69

• “NAT Support Parameters section” section on page 70

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SIP page

Voice tab > SIP page >

SIP Parameters section

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

SIP Server Name Server header used in responses to inbound responses.

SIP Reg User Agent Name

SIP Accept Language

DTMF Relay MIME Ty pe

Hook Flash MIME Ty pe

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.

The default is $VERSION.

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

The default is blank.

Accept-Language header used. There is no default (this indicates the WRP400 does not include this header). If empty, the header is not included.

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

The default is application/dtmf-relay.

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

SIP User Agent Name

parameter

Remove Last Reg Lets you remove the last registration before registering a

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The default is application/hook-flash.

new one if the value is different. Select yes or no from the drop-down menu.

The default is no.

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Use Compact Header

Escape Display Name

Lets you use compact SIP headers in outbound SIP messages. Select yes or no from the drop-down menu. If set to yes, the WRP400 uses compact SIP headers in outbound SIP messages. If set to no, the WRP400 uses normal SIP headers. If inbound SIP requests contain compact headers, the WRP400 reuses the same compact headers when generating the response regardless the settings of the SIP requests contain normal headers, the WRP400 substitutes those headers with compact headers (if defined by RFC 261) if set to yes.

The default is no.

Lets you keep the Display Name private. Select yes if you want the WRP400 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.

Use Compact Header

parameter. If inbound

parameter is

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

SIP TCP Port Min Specifies the lowest TCP port number that can be used for

SIP TCP Port Max Specifies the highest TCP port number that can be used

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 sessions.

for SIP sessions.

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Voice tab > SIP page >

SIP Timer Values (sec) section

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.

ReINVITE Expires ReINVITE request Expires header value. If you enter 0, the

Cisco Small Business WRP400 Administration Guide 65

–1 ) .

The default is 240. Range: 0–(2

Expires header is not included in the request.

The default is 30. Range: 0–(2

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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 WRP400 retries registration after

failing during the last registration.

The default is 30.

Reg Retry Long Intvl

Reg Retry Random Delay

Reg Retry Long Random Delay

Reg Retry Intvl Cap The maximum value to cap the exponential back-off retry

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

Retry Reg RSC

of time before retrying. If this interval is 0, the WRP400 stops trying. This value should be much larger than the Reg Retry Intvl value, which should not be 0.

The default is 1200.

Random delay range (in seconds) to add to

Intvl

when retrying REGISTER after a failure.

The default is 0, which disables this feature.

Random delay range (in seconds) to add to

Long Intvl

The default is 0, which disables this feature.

delay (which starts at every REGISTER retry after a failure). In other words, the retry interval is always at a failure. If this feature is enabled, is added on top of the exponential back-off adjusted delay value.

, the WRP400 waits for the specified length

when retrying REGISTER after a failure.

and doubles on

seconds after

Reg Retry Random Delay

Cisco Small Business WRP400 Administration Guide 66

The default value is 0, which disables the exponential backoff feature.

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Voice tab > SIP page >

Response Status Code Handling section

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

S I T4 To n e .

Try Backup RSC SIP response code that retries a backup server for the

current request.

Retry Reg RSC Interval to wait before the WRP400 retries registration after

failing during the last registration.

The default is 30.

Voice tab > SIP page >

RTP Parameters section

RTP Port Min Minimum port number for RTP transmission and reception.

RTP Port Min

The 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.

and

RTP Port Max

parameters should

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The default is 0.030.

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Max RTP ICMP Err Number of successive ICMP errors allowed when

transmitting RTP packets to the peer before the WRP400 terminates the call. If value is set to 0, the WRP400 ignores the limit on ICMP errors.

The default is 0.

RTCP Tx Interval Interval for sending out RTCP sender reports on an active

connection. It can range from 0 to 255 seconds. During an active connection, the WRP400 can be programmed to send out compound RTCP packet on the connection. Each compound RTP packet except the last one contains a SR (Sender Report) and a SDES.(Source Description). The last RTCP packet contains an additional BYE packet. Each SR except the last one contains exactly 1 RR (Receiver Report); the last SR carries no RR. The SDES contains CNAME, NAME, and TOOL identifiers. The CNAME is set to <User ID>@<Proxy>, NAME is set to <Display Name> (or Anonymous if user blocks caller ID), and TOOL is set to the Vendor/Hardware-platform-software-version (such as Cisco/wrp400-1.0.31(b)). The NTP timestamp used in the SR is a snapshot of the WRP400’s local time, not the time reported by an NTP server. If the WRP400 receives a RR from the peer, it attempts to compute the round trip delay and show it as the <Call Round Trip Delay> value (ms) in the Info section of the WRP400 Configuration Utility.

The default is 0.

No UDP Checksum Select yes if you want the WRP400 to calculate the UDP

header checksum for SIP messages. Otherwise, select no.

The default is no.

Stats In BYE Determines whether the WRP400 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.

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Voice tab > SIP page >

SDP Payload Types section

NSE Dynamic Payload

AV T D yna mi c Payload

INFOREQ Dynamic Payload

G726r16 Dynamic Payload

G726r24 Dynamic Payload

G726r32 Dynamic Payload

G726r40 Dynamic Payload

G729b Dynamic Payload

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

The default is 100.

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

The default is 101.

INFOREQ dynamic payload type.

There is no default.

G.726-16 dynamic payload type. The valid range is 96-127.

The default is 98.

G.726-24 dynamic payload type. The valid range is 96-127.

The default is 97. G726r32 dynamic payload type.

The default is 2. G.726-40 dynamic payload type. The valid range is 96-127.

The default is 96.

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

The default is 99.

NSE Codec Name NSE codec name used in SDP.

AV T C od ec Nam e AV T c od ec na me use d i n S DP.

G711u Codec Name

G711a Codec Name

G726r16 Codec Name

Cisco Small Business WRP400 Administration Guide 69

The default is NSE.

The default is telephone-event.

G.711u codec name used in SDP.

The default is computerMU.

G.711a codec name used in SDP.

The default is computerMA.

G.726-16 codec name used in SDP.

The default is G726-16.

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G726r24 Codec Name

G726r32 Codec Name

G726r40 Codec Name

G729a Codec Name

G729b Codec Name

G723 Codec Name G.723 codec name used in SDP.

EncapRTP Codec Name

G.726-24 codec name used in SDP.

The default is G726-24.

G.726-32 codec name used in SDP.

The default is G726-32.

G.726-40 codec name used in SDP.

The default is G726-40.

G.729a codec name used in SDP.

The default is G729a.

G.729b codec name used in SDP.

The default is G729ab.

The default is G723.

EncapRTP codec name used in SDP.

The default is EncapRTP.

Voice tab > SIP page >

NAT Support Parameters section

Handle VIA received

Handle VIA rport If you select yes, the WRP400 processes the rport

If you select yes, the WRP400 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.

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.

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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

STUN Enable Enables the use of STUN to discover NAT mapping. Select

STUN Test Enable If the STUN Enable feature is enabled and a valid STUN

STUN Server IP address or fully-qualified domain name of the STUN

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.

yes or no from the drop-down menu.

The default is no.

server is available, the WRP400 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 WRP400 detects symmetric NAT or a symmetric firewall, NAT mapping is disabled.

The default is no.

server to contact for NAT mapping discovery.

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EXT IP External IP address to substitute for the actual IP address

of the WRP400 in all outgoing SIP messages. If 0.0.0.0 is specified, no IP address substitution is performed.

If this parameter is specified, the WRP400 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 WRP400 is the edge device, the second requirement is met.

The default is 0.0.0.0.

Regional page

EXT RTP Port Min External port mapping number of the RTP Port Min.

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

The default is 0.

NAT Keep Alive Intvl

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” section on page 73

• “Distinctive Ring Patterns section” section on page 75

• “Distinctive Call Waiting Tone Patterns section” section on page 76

Interval between NAT-mapping keep alive messages.

The default is 15.

• “Distinctive Ring/CWT Pattern Names section” section on page 77

• “Ring and Call Waiting Tone Spec section” section on page 78

• “Control Timer Values (sec) section” section on page 78

• “Vertical Service Activation Codes section” section on page 80

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• “Outbound Call Codec Selection Codes section” section on page 86

• “Miscellaneous section” section on page 88

Voice tab > Regional page >

Call Progress Tones section

Dial Tone Prompts the user to enter a phone number. Reorder Tone is

played automatically when 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.

Dial Tone

or any of its

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

Off Hook Warning To n e

automatically when out.

The default is 480@-19,620@-19;10(.25/.25/1+2).

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

Dial Tone

or any of its alternatives times

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

Cisco Small Business WRP400 Administration Guide 73

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

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

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Ring Back 2 Tone Your WRP400 plays this ringback tone instead of

To n e

if the called party replies with a SIP 182 response without SDP to its outbound INVITE request. The default value is the same as 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

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

The default is 914@-16,1371@-16,1777@-16;20(.274/0/ 1,.274/0/2,.380/0/3,0/4/0).

Ring Back Tone

, except the cadence is

screen.

Ring Back

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

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

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).

screen.

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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

Feature Invocation To n e

Voice tab > Regional page >

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).

Played when a feature is implemented.

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

Distinctive Ring Patterns section

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(.3/.2, 1/.2,.3/4).

Ring3 Cadence Cadence script for distinctive ring 3.

Ring4 Cadence Cadence script for distinctive ring 4.

Ring5 Cadence Cadence script for distinctive ring 5.

Ring6 Cadence Cadence script for distinctive ring 6.

Ring7 Cadence Cadence script for distinctive ring 7.

Cisco Small Business WRP400 Administration Guide 75

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

The default is 60(.4/.2,.3/.2,.8/4).

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

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

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

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Ring8 Cadence Cadence script for distinctive ring 8.

The default is 60(0.25/9.75).

Voice tab > Regional page >

Distinctive Call Waiting Tone Patterns section

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, .1/9.3).

CWT4 Cadence Cadence script for distinctive CWT 4.

The default is 30(.1/.1, .3/ .1, .1/9.5).

CWT5 Cadence Cadence script for distinctive CWT 5.

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

CWT6 Cadence Cadence script for distinctive CWT 6.

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

CWT7 Cadence Cadence script for distinctive CWT 7.

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

CWT8 Cadence Cadence script for distinctive CWT 8.

The default is 2.3(.3/2).

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Voice tab > Regional page >

Distinctive Ring/CWT Pattern Names section

Ring1 Name Name in an INVITE’s Alert-Info Header to pick distinctive

ring/CWT 1 for the inbound call.

The default is Bellcore-r1.

Ring2 Name Name in an INVITE’s Alert-Info Header to pick distinctive

ring/CWT 2 for the inbound call.

The default is Bellcore-r2.

Ring3 Name Name in an INVITE’s Alert-Info Header to pick distinctive

ring/CWT 3 for the inbound call.

The default is Bellcore-r3.

Ring4 Name Name in an INVITE’s Alert-Info Header to pick distinctive

ring/CWT 4 for the inbound call.

The default is Bellcore-r4.

Ring5 Name Name in an INVITE’s Alert-Info Header to pick distinctive

ring/CWT 5 for the inbound call.

The default is Bellcore-r5.

Ring6 Name Name in an INVITE’s Alert-Info Header to pick distinctive

ring/CWT 6 for the inbound call.

The default is Bellcore-r6.

Ring7 Name Name in an INVITE’s Alert-Info Header to pick distinctive

ring/CWT 7 for the inbound call.

The default is Bellcore-r7.

Ring8 Name Name in an INVITE’s Alert-Info Header to pick distinctive

ring/CWT 8 for the inbound call.

The default is Bellcore-r8.

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Voice tab > Regional page >

Ring and Call Waiting Tone Spec section

IMPORTANT: Ring and Call Waiting tones don’t work the same way on all phones.

When setting ring tones, consider the following recommendations:

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

• If your ring cadence doesn’t sound right, or your phone doesn’t ring, change

your Ring Waveform, Ring Frequency, and Ring Voltage to the following:

- Ring Waveform: Sinusoid

- Ring Frequency: 25

- Ring Voltage: 80V

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.

Voice tab > Regional page >

Control Timer Values (sec) section

Hook Flash Timer Min

Hook Flash Timer Max

Minimum on-hook time before off-hook qualifies as hookflash. Less than this the on-hook event is ignored. Range:

0.1–0.4 seconds.

The default is 0.1.

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

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The default is 0.9.

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Callee On Hook Delay

Reorder Delay Delay after far end hangs up before reorder tone is played.

Call Back Expires Expiration time in seconds of a call back activation. Range:

Call Back Retry Intvl

Call Back Delay Delay after receiving the first SIP 18x response before

Phone must be on-hook for at this time in sec before the WRP400 will tear down the current inbound call. It does not apply to outbound calls. Range: 0–255 seconds.

The default is 0.

0 = plays immediately, inf = never plays. Range: 0–255 seconds.

The default is 5.

0–65535 seconds.

The default is 1800.

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

The default is 30.

declaring the remote end is ringing. If a busy response is received during this time, the WRP400 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 0.5.

Interdigit Long Timer

Interdigit Short 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.

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.

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Ccomputer Delay Delay in seconds after caller hangs up when the WRP400

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 Ccomputer feature should be used instead.

Without Ccomputer enabled, reorder tone will is played after a configurable delay. If Ccomputer is enabled, dial tone will be played when tip-to-ring voltage is restored Resolution is 1 second.

The default is 2.

Ccomputer Duration

Voice tab > Regional page >

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. Ccomputer is disabled if this value is set to 0. Range: 0 to 1.000 second. Resolution is

0.001 second.

The default is 0 (Ccomputer 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 dial-plan, although no harm is done if they are included.

Call Return Code This code calls the last caller.

The default is *69.

Call Redial Code Redials the last number called. .

Blind Transfer Code Begins a blind transfer of the current call to the extension

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The default is *07.

specified after the activation code.

The default is *98.

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Call Back Act Code Starts a callback when the last outbound call is not busy.

The default is *66.

Call Back Deact Code

Call Back Busy Act Code

Cfwd All Act Code Forwards all calls to the extension specified after the

Cfwd All Deact Code

Cfwd Busy Act Code

Cfwd Busy Deact Code

Cfwd No Ans Act Code

Cancels a callback.

The default is *86.

Starts a callback when the last outbound call is busy.

The default is *05

activation code.

The default is *72.

Cancels call forwarding of all calls.

The default is *73.

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

The default is *90.

Cancels call forwarding of busy calls.

The default is *91.

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

The default is *92.

Cfwd No Ans Deact Code

Cfwd Last Act Code

Cfwd Last Deact Code

Block Last Act Code

Block Last Deact Code

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Cancels call forwarding of no-answer calls.

The default is *93.

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

The default is *63.

Cancels call forwarding of the last inbound or outbound calls.

The default is *83.

Blocks the last inbound call.

The default is *60.

Cancels blocking of the last inbound call.

The default is *80.

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Accept Last Act Code

Accept Last Deact Code

CW Act Code Enables call waiting on all calls.

CW Deact Code Disables call waiting on all calls.

CW Per Call Act Code

CW Per Call Deact Code

Block CID Act Code Blocks caller ID on all outbound calls.

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.

Cancels the code to accept the last outbound call.

The default is *84.

The default is *56.

The default is *57.

Enables call waiting for the next call.

The default is *71.

Disables call waiting for the next call.

The default is *70.

The default is *67.

Block CID Deact Code

Block CID Per Call Act Code

Block CID Per Call Deact Code

Block ANC Act Code

Block ANC Deact Code

DND Act Code Enables the do not disturb feature.

DND Deact Code Disables the do not disturb feature.

Removes caller ID blocking on all outbound calls.

The default is *68.

Blocks caller ID on the next outbound call.

The default is *81.

Removes caller ID blocking on the next inbound call.

The default is *82.

Blocks all anonymous calls.

The default is *77.

Removes blocking of all anonymous calls.

The default is *87.

The default is *78.

The default is *79.

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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

Dist Ring Act Code Enables the distinctive ringing feature.

Dist Ring Deact Code

Speed Dial Act Code

Secure All Call Act Code

Secure No Call Act Code

Secure One Call Act Code

Disables call waiting, caller ID generation.

The default is *45.

The default is *26

Disables the distinctive ringing feature.

The default is *46.

Assigns a speed dial number.

The default is *74.

Makes all outbound calls secure.

The default is *16.

Makes all outbound calls not secure.

The default is *17.

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

Conference Act Code

Cisco Small Business WRP400 Administration Guide 83

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

The default is *19.

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.

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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

FAX Line Toggle Code

Referral Services Codes

Toggles the line to a modem.

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

Toggles the line to a fax machine.

The default is #99.

These codes tell the WRP400 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 WRP400 to perform a blind transfer to a target number that is prepended by the service *code.

For example, after the user dials *98, the WRP400 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 WRP400 sends a blind REFER to the holding party with the Refer-To target equals to *98 hand off a call to an application server to perform further processing, such as call park.

target_number

. This feature allows the WRP400 to

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The *codes should not conflict with any of the other vertical service codes internally processed by the WRP400. You can empty the corresponding *code that you do not want the WRP400 to process.

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Feature Dial Services Codes

These codes tell the WRP400 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 WRP400 to call the target number prepended by the *code. For example, after user dials *72, the WRP400 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 WRP400 sends a INVITE to *72 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 WRP400. You can empty the corresponding *code that you do not want to the WRP400 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)

target_number

as in a normal call. This

‘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 WRP400 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 WRP400 send INVITE *73@..... as usual when

user dials *73.

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Voice tab > Regional page >

Outbound Call Codec Selection Codes section

These codes automatically appended to the dial-plan. So no need to include them in dial-plan (although no harm to do so either).

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 G723 Code Makes this codec the preferred codec for the associated

call.

The default is *01723.

Force G723 Code Makes this codec the only codec that can be used for the

associated call.

The default is *02723.

Prefer G726r16 Code

Force G726r16 Code

Prefer G726r24 Code

Makes this codec the preferred codec for the associated call.

The default is *0172616.

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

The default is *0272616.

Makes this codec the preferred codec for the associated call.

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The default is *0172624.

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Force G726r24 Code

Prefer G726r32 Code

Force G726r32 Code

Prefer G726r40 Code

Force G726r40 Code

Prefer G729a Code Makes this codec the preferred codec for the associated

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

The default is *0272624.

Makes this codec the preferred codec for the associated call.

The default is *0172632.

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

The default is *0272632.

Makes this codec the preferred codec for the associated call.

The default is *0172640.

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

The default is *0272640.

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.

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Voice tab > Regional page >

Miscellaneous section

Set Local Date (mm/dd)

Set Local Time (HH/ mm)

Time Zone Selects the number of hours to add to GMT to generate the

FXS Port Impedance

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

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

local time for caller ID generation. Choices are GMT-12:00, GMT-11:00,…, GMT, GMT+01:00, GMT+02:00, …, GMT+13:00.

The default is GMT-08:00.

Sets the electrical impedance of the FXS port. Choices are

600, 900, 600+2.16uF, 900+2.16uF, 270+750||150nF, 220+850||120nF, 220+820||115nF, or 200+600||100nF.

The default is 600.

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Daylight Saving Time Rule

Enter the rule for calculating daylight saving time; it should include the start, end, and save values. This rule is comprised of three fields. Each field is separated by ; (a semicolon) as shown below. Optional values inside [ ] (the brackets) are assumed to be 0 if they are not specified. Midnight is represented by 0:0:0 of the given date.

SYNTAX: Start = <start-time>; end=<end-time>; save = <save-time>.

The <start-time> and <end-time> values specify the start and end dates and times of daylight saving time. Each value is in this format: <month> /<day> / <weekday>[/ HH:[mm[:ss]]]

The <save-time> value is the number of hours, minutes, and/or seconds to add to the current time during daylight saving time. The <save-time> value can be preceded by a negative (-) sign if subtraction is desired instead of addition. The <save-time> value is in this format: [/[+|-]HH:[mm[:ss]]]

The <month> value equals any value in the range 1-12 (January-December).

The <day> value equals [+|-] any value in the range 1-31.

If <day> is 1, it means the <weekday> on or before the end of the month (in other words the last occurrence of < weekday> in that month).

The <weekday> value equals any value in the range 1-7 (Monday-Sunday). It can also equal 0. If the <weekday> value is 0, this means that the date to start or end daylight saving is exactly the date given. In that case, the <day> value must not be negative. If the <weekday> value is not 0 and the <day> value is positive, then daylight saving starts or ends on the <weekday> value on or after the date given. If the <weekday> value is not 0 and the <day> value is negative, then daylight saving starts or ends on the <weekday> value on or before the date given.

The abbreviation HH stands for hours (0-23).

The abbreviation mm stands for minutes (0-59).

The abbreviation ss stands for seconds (0-59).

The default Daylight Saving Time Rule is start=4/1/ 7;end=10/-1/7;save=1.

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Daylight Saving Time Enab le

FXS Port Input Gain Input gain in dB, up to three decimal places. The range is

FXS Port Output Gain

DTMF Playback Level

DTMF Playback Length

Detect ABCD To enable local detection of DTMF ABCD, select yes.

Daylight Saving Time can be turned on or off. This option affects the time stamp on CallerID and affects all the lines and extensions of the phone. Default is Yes (on).

6.000 to -12.000.

The default is -3.

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

parameter.

Gain

The default is -3.

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

The default is -16.0.

Local DTMF playback duration in milliseconds.

The default is .1.

Otherwise, select no.

FXS Port Output

The default is yes. Setting has no effect if DTMF Tx Method is INFO; ABCD is always sent OOB regardless in this setting.

Playback ABCD To enable local playback of OOB DTMF ABCD, select yes.

Otherwise, select no.

The default is yes.

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Caller ID Method The following choices are available:

Bellcore (N.Amer,China)—CID, CIDCW, and VMWI. FSK sent

•

after first ring (same as ETSI FSK sent after first ring) (no polarity reversal or DTAS).

• DTMF (Finland, Sweden)—CID only. DTMF sent after polarity

reversal (and no DTAS) and before first ring.

• DTMF (Denmark)—CID only. DTMF sentbefore first ring with

no polarity reversal and no DTAS.

• ETSI DTMF—CID only. 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).

Caller ID FSK Standard

Feature Invocation Method

More Echo Suppression

• 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.

• DTMF (Denmark) With PR—CID only. DTMF sent after

polarity reversal (and no DTAS) and before first ring.

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

The WRP400 supports bell 202 and v.23 standards for caller ID generation. Select the FSK standard you want to use, bell 202 or v.2 3.

The default is bell 202.

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

Enable or disable more echo suppresion. The default is no.

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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” section on page 92

• “Streaming Audio Server (SAS) section” section on page 93

• “NAT Settings section” section on page 94

• “Network Settings section” section on page 94

• “SIP Settings section” section on page 95

• “Call Feature Settings section” section on page 98

• “Proxy and Registration section” section on page 99

• “Subscriber Information section” section on page 101

• “Supplementary Service Subscription section” section on page 102

• “Audio Configuration section” section on page 104

• “Dial Plan section” section on page109

• “FXS Port Polarity Configuration section” section on page 110

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.

Voice tab > Line page >

Line Enable section

Line Enable

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

The default is yes.

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Line page

Voice tab > Line page >

Streaming Audio Server (SAS) section

SAS Enable

SAS DLG Refresh Intvl

SAS Inbound RTP Sink

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.

If this value is not zero, it is the interval at which the streaming audio server sends out session refresh (SIP reINVITE) messages to determine whether the connection to the caller is still active. If the caller does not respond to the refresh message, the WRP400 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.

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.

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Line page

Voice tab > Line page >

NAT Settings section

NAT Mapping Enable

NAT Keep Alive Enable

NAT Keep Alive Msg

NAT Keep Alive Dest

Voice tab > Line page >

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

The default is no.

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

The default is no.

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.

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

SIP ToS/DiffServ Value

SIP CoS Value [0-7]

RTP ToS/DiffServ Value

RTP CoS Value [07]

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

The default is 0x68.

CoS value for SIP messages.

The default is 3.

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

The default is 0xb8.

CoS value for RTP data.

The default is 6.

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Network Jitter Level

Jitter Buffer Adjustment

Voice tab > Line page >

Determines how jitter buffer size is adjusted by the WRP400. Jitter buffer size is adjusted dynamically. The minimum jitter buffer size is 30 milliseconds or (10 milliseconds + current RTP frame size), whichever is larger, for all jitter level settings. However, the starting jitter buffer size value is larger for higher jitter levels. This setting controls the rate at which the jitter buffer size is adjusted to reach the minimum. Select the appropriate setting: low, medium, high, very high, or extremely high.

The default is high.

Controls how the jitter buffer should be adjusted. Select the appropriate setting: up and down, up only, down only, or disable.

The default is up and down.

SIP Settings section

Field Description

SIP Transport

SIP Port

SIP 100REL Enable

EXT SIP Port The external SIP port number.

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.

Port number of the SIP message listening and transmission port.

The default is 5060.

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.

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Auth ResyncReboot

SIP Proxy-Require

SIP Remote-PartyID

SIP GUID

If this feature is enabled, the WRP400 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.

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.

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

The default is yes.

The Global Unique ID is generated for each line for each device. When it is enabled, the WRP400 adds a GUID header in the SIP request. The GUID is generated the first time the unit boots up and stays with the unit through rebooting and even factory reset. This feature was requested by Bell Canada (Nortel) to limit the registration of SIP accounts.

The default is yes.

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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

• 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. 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 WRP400 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

is yes).

(or

Outbound Proxy

Use Outbound

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The default is no.

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Referor Bye Delay

Refer Target Bye Delay

Referee Bye Delay

R e fe r -To Tar ge t Contact

Sticky 183

Controls when the WRP400 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.

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

The default is 0.

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

The default is 0.

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

The default is no.

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

Voice tab > Line page >

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

Call Feature Settings section

Blind Attn-Xfer Enable

Enables the WRP400 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 WRP400 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.

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