Cisco Spa504g Administration Guide

ADMINISTRATION GUIDE
Cisco Small Business IP Phones Models SPA301, 303, 501G, 502G, 504G, 508G, 509G, 525G/525G2, and WIP310
Document revised December 2010
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2010 Cisco Systems, Inc. All rights reserved.
OL-19749-05
Contents
Chapter 1: Getting Started

Overview of the Phones Cisco SPA500S Attendant Console Network Configurations
Cisco SPA 9000 Voice System Cisco Unified Communications 500 Series for Small Business Other SIP IP PBX Call Control Systems
12
12 14 14
15 16 17
Prerequisites Upgrading Firmware

Determining the Current Firmware Version Determining Your IP Address Downloading the Firmware Installing the Firmware
17 18
18 20 21 21
Using the Web-Based Configuration Utility

Understanding Administrator and User Views Accessing Administrative Options Understanding Basic and Advanced Views Using the Web Administration Tabs Roadmap to Configuration Utility Features
22
24 25 25 25 25
Viewing Phone Information Using IVR on the Cisco SPA301 and Cisco SPA501G IP Phone
29 29
Chapter 2: Configuring Lines and Extensions

Configuring Lines
Shared Line Appearances Configuring a Line Configuring Shared Line Appearance Assigning Busy Lamp Field, Call Pickup, and Speed Dial Functions to Unused Lines on a Cisco SPA300 Series or Cisco SPA500 Series IP Phone
Configuring Call Pickup and Busy Lamp Field Configuring Speed Dial
33
33
33 34 35
36
36 38
Cisco Small Business SPA300 Series, SPA500 Series, and WIP310 IP Phone Administration Guide
Contents
Configuring Unused Line Keys for Call Park on the Cisco SPA525G/525G2 (MetaSwitch) Configuring Unused Line Keys to Access Services Configuring Line Key LED Patterns on the Cisco SPA300 Series or Cisco SPA500 Series IP Phone
39 40 41
Configuring Extensions
43
Chapter 3: Customizing Cisco SPA and Wireless IP Phones

Configuring Phone Information and Display Settings
Configuring the Phone Name Configuring Voice Mail
Configuring Internal Voice Mail for Each Extension (When Using Cisco SPA 400 for Voice Mail)
45
46
46 46
47
Customizing the Startup Screen Changing the Display Background (Cisco SPA300 Series and Cisco SPA500 Series) Configuring the Screen Saver Configuring the LCD Contrast Configuring Back Light Settings (Cisco SPA525G/525G2)
47 49 51 52 53
Configuring Linksys Key System Parameters Enabling Call Features

Enabling Anonymous Call and Caller ID Blocking Services Enabling Automatic Call Distribution (ACD) Service Enabling Call Back Service Enabling Call Park and Call Pickup Services Enabling Call Transfer and Call Forwarding Services Enabling Conferencing Enabling Do Not Disturb Enabling the Missed Call Shortcut Logging Missed Calls (Cisco SPA300 Series and Cisco SPA500 Series)
54 54
55 55 56 56 57 57 58 58 59
Contents
Enabling Paging (Intercom)

Single Page Group Paging Configuring a Phone to Automatically Accept Pages Configuring Paging Groups
59
59 59 60 60
Enabling Secure Call Enabling Service Announcements
62 62
Customizing Phone Softkeys

Programmable Softkeys Configuration Example
62
67 68
Configuring the Message Waiting Indicator Configuring Ring Tones

Configuring On-Demand Ring Tones (Cisco SPA525G/525G2) User-Created MP3 Ring Tones (Cisco SPA525G/525G2) Creating and Uploading Ring Tones Using the Ring Tone Utility (Cisco SPA30X and Cisco SPA50XG only) Assigning a Ring Tone to an Extension
69 70
71 72 72 73
Configuring RSS Newsfeeds on the Cisco SPA525G/525G2 IP Phone Configuring Audio Settings
Configuring Audio Input Gain (Cisco SPA300 Series and Cisco SPA500 Series)
74 75
76
Enabling Wireless (Cisco SPA525G/525G2 only) Configuring Bluetooth (Cisco SPA525G/525G2 only)
Enabling Bluetooth (Cisco SPA525G/525G2) Using a Bluetooth Headset (Cisco SPA525G/525G2) Pairing Your Cisco SPA525G2 with a Bluetooth-Enabled Mobile Phone
Initiating Pairing from the Cisco SPA525G2 Initiating Pairing from Your Bluetooth-Enabled Mobile Phone
77 77
77 78 80
80 81
Enabling SMS Messaging Enabling the Web Server Configuring Lightweight Directory Access Protocol (LDAP) for the Cisco SPA300 Series and Cisco SPA500 Series IP Phones
82 84 85
Contents
Configuring BroadSoft Settings (Cisco SPA300 Series and Cisco SPA500 Series)
Configuring BroadSoft Directory Configuring Synchronization of Do Not Disturb and Call Forward
89
89 90
Configuring XML Services Configuring Music On Hold Configuring Extension Mobility Configuring Video Surveillance on the Cisco SPA525G/525G2
Configuring the User Name and Account on the Camera Entering Camera Information Into the Cisco SPA525G/525G2 Configuration Utility Viewing the Video
91 93 94 95
96 96 97
Chapter 4: Configuring SIP, SPCP, and NAT

Session Initiation Protocol and Cisco IP Phones
SIP Over TCP SIP Proxy Redundancy
Configuring Survivable Remote Site Telephony (SRST) Support
98
98
99 100
100
RFC3311 Support Support for SIP NOTIFY XML-Service
101 101
Configuring SIP
Configuring SIP Parameters Configuring SIP Timer Values Configuring Response Status Code Handling Configuring RTP Parameters Configuring SDP Payload Types Configuring SIP Settings for Extensions
Configuring a SIP Proxy Server Configuring Subscriber Information Parameters
101
101 105 108 108 110 113
118 121
Configuring SPCP on the Cisco SPA525G/525G2 Configuring SPCP on the Cisco SPA300 Series and Cisco SPA50XG Network Address Translation (NAT) and Cisco IP Phones
122 123 123
Contents
NAT Mapping with Session Border Controller NAT Mapping with SIP-ALG Router Configuring NAT Mapping with a Static IP Address Configuring NAT Mapping with STUN Determining Whether the Router Uses Symmetric or Asymmetric NAT
124 124 124 125 126
Chapter 5: Configuring Security, Quality, and Network Features

Setting Security Features
SIP Initial INVITE and MWI Challenge SIP Over TLS SRTP and Securing Calls
Secure Call Indication Tone
128
128
129 129 130
131
Ensuring Voice Quality

Supported Codecs Bandwidth Requirements Factors Affecting Voice Quality
132
132 133 134
Configuring Voice Codecs Configuring Domain and Internet Settings

Configuring Restricted Access Domains Configuring DHCP, Static IP, and PPPoE Information
Setting a Static IP Address Configuring PPPoE Settings (Cisco SPA525G/525G2 only)
136 140
140 140
141 142
Setting Optional Network Parameters Configuring VLAN Settings for IP Phones

Configuring Cisco Discovery Protocol (CDP) Configuring Link Layer Discovery Protocol for Media Endpoint Devices (LLDP-MED)
TLV Information
143 145
145 146
147
Configuring the VLAN Settings
154
Configuring SSL VPN on the Cisco SPA525G/525G2

Configuring the VPN on the Security Appliance Configuring the VPN on the Cisco SPA525G/525G2
155
156 157
Contents
Chapter 6: Provisioning Basics

Provisioning Capabilities
Provisioning Configuration from Phone Keypad
160
161
161
IP Phone Configuration Profiles

Obtaining the SPC Tool General Purpose Parameters
163
164 165
Sample Configuration File Upgrading, Resyncing, and Rebooting Phones

Firmware Upgrade Parameters Resyncing a Phone Rebooting a Phone
165 166
167 168 169
Redundant Provisioning Servers Retail Provisioning Automatic In-House Preprovisioning Configuration Access Control
Restricting User Access to the Phone Interface Menus (Cisco SPA300 and Cisco SPA500 Series)
169 169 170 171

171
Using HTTPS
Server Certificates Client Certificates Obtaining a Server Certificate
172
173 173 173
Chapter 7: Configuring Regional Parameters and Supplementary Services

Advanced Scripting for Cadences, Call Progress Tones, and Ring Tones
Example 1: Normal Ring Example 2: Distinctive Ring (short,short,short,long) Example 1: Dial Tone Example 3: SIT Tone Example 1: SIT Tone
175
176
176 176 177 177 178
Call Progress Tones
179
Contents
Distinctive Ring Patterns Control Timer Values (sec) Configuring Supplementary Services (Star Codes)
Entering Star Code Values Activating or Deactivating Supplementary Services
179 180 181

181 186
Vertical Service Announcement Codes (Cisco SPA300 and Cisco SPA500 Series)
Bonus Services Announcement description Outbound Call Codec Selection Codes
186
187 188
Miscellaneous Parameters
DTMF Parameters
189
189
Localizing Your IP Phone

Managing the Time and Date Configuring Daylight Saving Time
Daylight Saving Time Examples
190
191 192
193
Selecting a Display Language Creating a Dictionary Server Script
193 194
Chapter 8: Configuring Dial Plans

About Dial Plans
Digit Sequences Digit Sequence Examples Acceptance and Transmission of the Dialed Digits Dial Plan Timer (Off-Hook Timer)
Syntax for the Dial Plan Timer
196
196
198 200 202 203
203
Interdigit Long Timer (Incomplete Entry Timer)

Syntax for the Interdigit Long Timer
204
204
Interdigit Short Timer (Complete Entry Timer)

Syntax for the Interdigit Short Timer
205
205
Editing Dial Plans on the IP Phone Resetting the Control Timers
206 207
Contents
Chapter 9: Configuring the Cisco SPA500S Attendant Console

Cisco SPA500S Features Setting Up the Cisco SPA500S Attendant Console
Configuring the Cisco SPA 9000 for the Cisco SPA500S Configuring the BroadSoft Server for the Cisco SPA500S Configuring the Asterisk Server for the Cisco SPA500S Configuring the Cisco SPA500S Unit/Key Configuration Scripts
Assigning Cisco SPA500S LEDs to Phone Extensions Cisco SPA 9000 Syntax BroadSoft syntax Asterisk syntax
208
209 210
211 211 212 213 214
215 216 216 216
Configuring BroadSoft Busy Lamp Field Auto-Configuration Attendant Console Parameters
217 217
Monitoring the Cisco SPA500S

Cisco SPA500S Unit Monitoring Notes
219
219
Chapter A: Creating an LED Script

LED Script Examples LED Pattern
220
221 221
Appendix B: Cisco SPA IP Phone Field Reference

Info Tab
System Information Network Configuration (SPCP) VPN Status (SPA525G/525G2 Only) Product Information Phone Status Ext Status Line/Call Status Downloaded Ring Tone
223
224
224 226 227 227 228 230 230 232
System Tab
System Configuration
233
233
Contents
Internet Connection Type and Static IP Settings PPPoE Settings (Cisco SPA525G/525G2 Only) Optional Network Configuration VLAN Settings Wi-Fi Settings (Cisco SPA525G/525G2 Only) Bluetooth Settings (Cisco SPA525G/525G2 Only) VPN Settings (Cisco SPA525G/525G2 Only)
235 236 236 237 239 239 239
SIP Tab
SIP Parameters SIP Timer Values (sec) Response Status Code Handling RTP Parameters SDP Payload Types NAT Support Parameters Linksys Key System Parameters
240
240 244 246 247 249 252 254
Provisioning Tab Regional Tab

Call Progress Tones Distinctive Ring Patterns Control Timer Values (sec) Vertical Service Activation Codes Vertical Service Announcement Codes Outbound Call Codec Selection Codes Time (Cisco SPA525G/525G2 Only) Language (Cisco SPA525G/525G2 Only) Miscellaneous
254 255
255 258 259 260 265 265 268 268 268
Phone Tab
General Line Key Miscellaneous Line Key Settings Line Key LED Pattern Supplementary Services
274
274 277 279 279 281
Contents
Ring Tone (Cisco SPA300 Series and Cisco SPA500 Series) Ring Tone (WIP310) Auto Input Gain (dB) Multiple Paging Group Parameters BroadSoft Settings Lightweight Directory Access Protocol (LDAP) Corporate Directory Search XML Service Extension Mobility Programmable Softkeys
283 284 285 286 286 288 291 291 292
Ext Tab
General Share Line Appearance NAT Settings Network Settings SIP Settings Call Feature Settings Proxy and Registration Subscriber Information Audio Configuration Dial Plan
294
295 295 296 296 297 301 304 306 307 310
User Tab
Call Forward Speed Dial Supplementary Services Camera Settings (Cisco SPA525G/525G2) Web Information Service Settings (Cisco SPA525G/525G2) Audio Volume Screen (Cisco SPA525G/525G2)
311
312 312 313 313 313 313 314
Attendant Console Tab (Cisco SPA500 Series)

General Unit 2
316
316 317
10
Contents
Attendant Console Status Cisco SPA525G/525G2-Specific Tabs

Wi-Fi Bluetooth Personal Address Book Call History Speed Dials Firmware Upgrade
317 318
318 319 320 321 321 321
Appendix C: Where to Go From Here
322
11
1
Getting Started
This chapter contains basic information on Cisco SPA300 Series, Cisco SPA500 Series and Cisco Wireless-G IP Phones. It includes the following sections: Overview of the Phones, page 12 Network Configurations, page 14 Prerequisites, page 17 Upgrading Firmware, page 18 Using the Web-Based Configuration Utility, page 22 Viewing Phone Information, page 29 Using IVR on the Cisco SPA301 and Cisco SPA501G IP Phone, page 29
Overview of the Phones

The Cisco SPA IP Phone family is a line of full-featured VoIP (Voice over Internet Protocol) phones that provide voice communication over an IP network. They provide all the features of traditional business phones, such as call forwarding, redialing, speed dialing, transferring calls, conference calling and accessing voice mail. Calls can be made or received with a handset, a headset, or a speaker.
12
Getting Started
Overview of the Phones
1
The Cisco SPA IP Phone family includes the models shown in the following table: Model
SPA301 SPA303
Screen
No 128 X 64 monochrome LCD Paper labels
Lines
1 3
Softkeys
None 4 dynamic
Navigation Button
No Four-way navigation key
SPA501G
4 fixed (forward, cancel, conference, and transfer) 4 dynamic
No
SPA502G SPA504G SPA508G SPA509G SPA525G/ SPA525G2
128 X 64 monochrome LCD with backlight
1 4 8 12
Four-way navigation key
320 X 240 color highresolution LCD with backlight 128 X 160 color with backlight
Four-way navigation key with centerselect button
WIP310
None
For more information on phone features, see the data sheets for each product.
13
Getting Started
Cisco SPA500S Attendant Console
Cisco SPA500S Attendant Console

The Cisco SPA500S Attendant Console is used with the Cisco SPA300 Series and Cisco SPA500 Series IP Phone models to provide additional lines. The Cisco SPA500S has 32 LEDs/buttons for dialing, call transfer, call pick up and call monitoring. Multi-colored LEDs monitor the status of each configured voice line via busy lamp field (BLF). You can attach two attendant consoles to an IP phone, for 64 LEDs/buttons. For more information, see Chapter 9, Configuring the Cisco SPA500S Attendant Console.
Network Configurations
The Cisco SPA IP Phones support Session Initiation Protocol (SIP) or Smart Phone Control Protocol (SPCP). SPCP is supported only on the Cisco SPA300 Series or Cisco SPA500 Series IP Phones.) You can use the Cisco SPA IP Phones as part of a Cisco SPA 9000 Voice System phone network, or with any vendors IP PBX system that supports SIP. The Cisco SPA IP phones can be used as part of a Cisco SPA 9000 Voice System phone network, a SIP network, or as part of the Cisco Unified Communications 500 Series for Small Business.
14
Getting Started
1
SPA IP Phones IP WIP310
PSTN Gateway
Smart Switch
IP PBX
Wireless Router Analog Phone Lines Integrated Access Device
Analog Phones or Fax Machines
IAD
Internet
Figure 1 Network Configuration Example

NOTE Using the Cisco SPA IP Phones as part of a Cisco SPA 9000 Voice System Network
provides seamless integration of advanced features, such as paging, call pickup, and shared line appearances. This document describes some common network configurations; however, your configuration may vary depending on the type of equipment used by your service provider.
Cisco SPA 9000 Voice System

The Cisco SPA 9000 Voice System is an affordable and feature-rich voice over IP (VoIP) telephone system that is designed especially for small businesses. The Cisco SPA 9000 Voice System uses standard TCP/IP protocols and can provide global connectivity through any Internet Telephony Service Provider (ITSP) that supports SIP.
276459
15
Getting Started
1
At minimum, the Cisco SPA 9000 Voice System includes a Cisco SPA 9000 IP PBX and one or more Cisco SPA IP Phones. These devices are connected through a switch to a local area network. With an Internet connection, the Cisco SPA 9000 Voice System can subscribe to ITSP services to take advantage of low calling rates. With the optional Cisco SPA 400, the Cisco SPA 9000 Voice System can connect to the Public Switched Telephone Network (PSTN) to support legacy phone lines. The Cisco SPA 400 also provides local voice mail service. When you use Cisco SPA IP Phones with the Cisco SPA 9000 Voice System, the following additional phone features are available: Auto attendant for multiple extensions Music on hold Configurable call routing Multiple DID numbers per VoIP line Call hunting (sequential, round robin, random) Group paging Call parking Call pick up Group call pick up
You can configure and manage the Cisco SPA 9000 Voice System using an Interactive Voice Response (IVR) system, the Cisco SPA 9000 Voice System Setup Wizard, or a built-in web server. For more information, see the Cisco SPA 9000
Voice System Administration Guide.
Cisco Unified Communications 500 Series for Small Business

The Cisco Unified Communications 500 Series for Small Business is an affordable appliance that provides voice, data, voice mail, Automated Attendant, video, security, and wireless capabilities while integrating with existing desktop applications such as calendar, e-mail, and customer relationship management (CRM) programs. The Cisco SPA300 Series and Cisco SPA500 Series IP Phones can be configured to work with this system. This system uses Smart Phone Control Protocol (SPCP).
16
Getting Started
Prerequisites
1
Other SIP IP PBX Call Control Systems
The Cisco SPA IP Phones are compatible with other IP PBX call control systems, such as BroadSoft and Asterisk, that use SIP for call processing. Configuration of those systems is not covered in this document. Additional resources for configuring the Cisco SPA IP Phones to work with these systems are available in Appendix C, Where to Go From Here.
Prerequisites
This document assumes that you have performed the following prerequisites before administering your Cisco SPA IP Phones. If you have not completed these prerequisites, see the documentation in Appendix C, Where to Go From Here, for more information. 1. Set up your IP network. 2. Configure the wireless network (required for Cisco SPA525G/525G2 and WIP310). 3. Install and configure the call control system, such as such as a Cisco SPA Cisco SPA 9000, Cisco Unified Communications 500 Series for Small Business, or an Internet-based IP PBX. 4. Update firmware. See Upgrading Firmware, page 18.
17
Getting Started
Upgrading Firmware
1
Phones should be upgraded to the latest firmware before using any administration features. There are various ways to upgrade your firmware: All Phones Cisco SPA 9000 Voice System Setup WizardIf you are using the Cisco SPA IP Phones with a Cisco SPA 9000, you can use the Cisco SPA 9000 Voice System Setup Wizard to upgrade your phones. See the Cisco SPA 9000 Voice System Setup Wizard User Guide. (Note: The Setup Wizard does not support the Cisco SPA300 Series IP Phones.) AutoprovisioningA configuration file that includes upgrade information is downloaded by a users phone when it is powered on. See the Upgrading, Resyncing, and Rebooting Phones section on page 166.
Upgrading Firmware
Cisco SPA30X, Cisco SPA50XG, and WIP310 Firmware Upgrade Executable File (Cisco SPA30X, Cisco SPA50X or WIP310)Download the firmware upgrade utility from the product page on Cisco.com to your PC desktop and run the upgrade from your PC by double-clicking the executable file. Your computer must be on the same network as the Cisco SPA IP Phone.
Cisco SPA525G/525G2 Configuration Utility (Cisco SPA525G/525G2)You can download the latest firmware onto your PC desktop and use the configuration utility to upgrade your firmware.
WIP310 TFTP/HTTP serverThe latest firmware image file is loaded onto an HTTP/ TFTP server and is accessed by a web browser. See the Cisco WIP310 User Guide for more information.
Determining the Current Firmware Version

Before upgrading, determine the current firmware version:
18
Getting Started
Upgrading Firmware
1
Cisco SPA301G
STEP 1 Quickly press the asterisk (*) button four times. STEP 2 In the IVR menu, enter 150, then press #. The firmware version is recited.
Cisco SPA501G
STEP 1 Press the Setup button. STEP 2 In the IVR menu, enter 150, then press #. The firmware version is recited.
Cisco SPA303, Cisco SPA502G, Cisco SPA504G, Cisco SPA508G, Cisco SPA509G
STEP 1 Press the Setup button. STEP 2 Scroll to Product Info and then press Select. The current firmware is displayed
under Software Version.
Cisco SPA525G/525G2
STEP 1 Press the Setup button. STEP 2 Scroll to Status and press Select. STEP 3 Select Product Information. The current firmware is displayed under Software
Version.
WIP310
STEP 1 In the Home screen, press the Options, highlight Phone Info, and press the Select
button.
STEP 2 Scroll to Software Version.
19
Getting Started
Upgrading Firmware
1
Determining Your IP Address
Before you upgrade, youll need the IP address of the phone you are upgrading. To get your IP address: Cisco SPA301
STEP 1 Quickly press the asterisk (*) button four times. STEP 2 In the IVR menu, enter 110, then press #. The IP address is recited.
Cisco SPA501G
STEP 1 Press the Setup button. STEP 2 In the IVR menu, enter 110, then press #. The IP address is recited.
Cisco SPA301, Cisco SPA502G, Cisco SPA504G, Cisco SPA508G, Cisco SPA509G
STEP 1 Press the Setup button. STEP 2 Scroll to Network and press Select. The IP Address is displayed under Current IP.
Cisco SPA525G/525G2
STEP 1 Press the Setup button. STEP 2 Scroll to Status and press Select. STEP 3 Scroll to Network Status and press Select. The IP address of your phone is
displayed.
WIP310
STEP 1 In the Home screen, press the Select key and navigate to Settings. STEP 2 Press the Select key and navigate to Phone Info.
20
Getting Started
Upgrading Firmware
STEP 3 The IP Address section displays the IP address.
Downloading the Firmware

NOTE Requires a Cisco.com user ID and password. STEP 1 Direct your browser to the following URL: http://www.cisco.com/public/sw-
center/index.shtml.
STEP 2 Search to locate your product. STEP 3 Locate the download page and download the firmware file. STEP 4 If the firmware file you download is in zip format, double-click the file and extract
its contents to a single folder or to the desktop.
Installing the Firmware

NOTE Your computer must be on the same sub-network as the phone you are upgrading.
Cisco SPA30X and Cisco SPA50XG

STEP 1 Run the executable file for the firmware upgrade. STEP 2 Click Continue after reading the message regarding upgrading and your service
provider.
STEP 3 Enter the IP address of your phone. STEP 4 Follow the on-screen directions.
Cisco SPA525G/525G2
STEP 1 Log in to the configuration utility for the phone. STEP 2 Choose the Firmware Upgrade tab. STEP 3 Click Firmware Upgrade Window.
21
Getting Started
STEP 4 Browse to select the firmware file from your PC. Click Submit. The firmware is
installed and your phone reboots.
WIP310
STEP 1 Turn off your WIP310 and connect it to your computer by using the USB cable. STEP 2 Double-click the executable file for the firmware upgrade (for example, double-
click wip310-5-0-11.exe).
STEP 3 Follow the on-screen instructions. STEP 4 When the upgrade is complete, disconnect the phone from your PC and power it
on.

You must be connected to the same network as your phone. For example, if you are connected to a VPN, you must first disconnect.
NOTE If you are using the Cisco SPA IP Phones with the Cisco Unified Communications
500 Series for Small Business for Call Control, use Cisco Unified Communication Manager Express or Cisco Configuration Assistant for phone administration. For more information, refer to the Cisco Unified Communications 500 Office Administrator Guide or the Cisco Configuration Assistant Smart Business Communications System Administrator Guide.
22
Getting Started
To access the IP phone configuration utility:

STEP 1 Launch a web browser on a computer that can reach the phone on the network. STEP 2 Direct the browser to the IP address of the phone. To determine the IP address:
Cisco SPA301: Press the asterisk (*) button four times. In the IVR menu, enter 110, then press #. The IP address is recited. Cisco SPA501G: Press the Setup button. In the IVR menu, enter 110, then press #. The IP address is recited. Cisco SPA303, Cisco SPA502G, Cisco SPA504G, Cisco SPA508G, Cisco SPA509G: Press the Setup button, then select Network. The Current IP field shows the phones current IP address. WIP310: In the Home screen, press Options and highlight Phone Info. Press the Select button. Cisco SPA525G/525G2: Press the Setup button, then select Status. Select Network Status. The IP address is displayed.
STEP 3 Enter the IP address in your web browser address bar. For example:
http://192.168.1.8
NOTE If your service provider disabled access to the configuration utility, you must
contact the service provider. If you have trouble accessing the configuration utility, perform the following steps: Cisco SPA303, Cisco SPA502G, Cisco SPA504G, Cisco SPA508G, Cisco SPA509G
STEP 1 Press the Setup button on the phone. STEP 2 Select Network. STEP 3 Scroll to Enable Web Server and make sure that it is set to Yes. If not, press the
Edit soft key and press y/n soft key to set it to Yes.
STEP 4 Press OK, then press Save.
23
Getting Started
Cisco SPA501G
STEP 1 Press the Setup button on the phone. STEP 2 In the IVR menu, enter 7932. STEP 3 Press 1 to enable the web server, then press #. STEP 4 To save, press 1; to review, press 2; to re-enter, press 3; to exit, press *.
WIP310
STEP 1 In the Home screen, press the Select button to choose Settings. STEP 2 Press the Select button again to reach the Settings menu. STEP 3 Scroll to highlight Misc Settings and press the Select button. STEP 4 Press the left arrow to ensure that Enable Web Server is set to On. STEP 5 Press the Select button to save this setting.
Cisco SPA525G/525G2
STEP 1 Press the Setup button. STEP 2 Select Network Configuration. STEP 3 Scroll to Web Server and make sure it is set to On. STEP 4 Press Save.
Understanding Administrator and User Views

Depending on whether you are a VAR or service provider, you might have different privileges. By default, the Administrator account name is admin, and the User account name is user. These account names cannot be changed. If the service provider set an Administrator account password, you are prompted for it when you click Admin Login.
24
Getting Started
The Administrator account can modify all web profile parameters, including web parameters available to the user login. The Administrator specifies the parameters that a User account can modify using the Provisioning tab of the configuration utility.
NOTE No default passwords are assigned to either the Administrator or User accounts.
Only the Administrator account can assign and change passwords.
Accessing Administrative Options

To access administrative options, either: Log in to the configuration utility, then click Admin Login. Enter the following URL when accessing the configuration utility: http://phone.ip.address/admin/
NOTE To save changes on a web page, click Submit All Changes before switching
between User and Admin Login or between basic and advanced views. Switching logins or views discards any unsubmitted changes.
Understanding Basic and Advanced Views

These views are similar, but advanced view shows more options on each web page. To see all available options for your login, use the advanced view.
Using the Web Administration Tabs

Each tab contains different parameters. Some tasks require you to set parameters in different tabs. For field reference about each parameter available on the configuration utility, see Appendix B, Cisco SPA IP Phone Field Reference.
Roadmap to Configuration Utility Features

The following tables provide a roadmap to features available in the configuration utility.
25
Getting Started
1
In the configuration utility, click the ... Info tab See Viewing Phone Information section on page 29.
To perform these tasks... View phone, extension, and line/call information, including:
DHCP, current IP address, DNS addresses Software and hardware versions Broadcast, RTP, and SIP information Registration state Packets sent, received, lost, and other information
Configure system-level parameters, including network and debug parameters. To:
System tab See Chapter 5, Configuring Security, Quality, and Network Features.
Enable the configuration utility and administrator access Set the Internet connection type to DHCP Configure the syslog and debug servers Enable VLAN and CDP
Configure parameters to adjust SIP stack and protocols. To enable:
SIP tab See Chapter 4, Configuring SIP, SPCP, and NAT.
CTI SIP-B STUN
Configure provisioning parameters. To:
Provisioning tab The Provisioning tab is viewable by Admin logins only. See Chapter 6, Provisioning Basics. For additional information about provisioning, see the Cisco Small Business IP
Enable remote provisioning Enable firmware upgrades Set general purpose parameters
Telephony Devices Provisioning Guide (for Cisco service

providers).
26
Getting Started
1
In the configuration utility, click the ... Regional tab See Chapter 7, Configuring Regional Parameters and Supplementary Services.
To perform these tasks... Configure parameters that depend on country or region, including:
Call progress tones Ring patterns Star codes/vertical service activation codes Vertical service announcement codes Local date/time and language
Configure General phone station info, which applies to all extensions configured for the phone, including:
Phone tab
See Chapter 3, Customizing Cisco SPA and Wireless IP Phones.
Station name, voice mail number, text logos and background pictures Extension numbers for line keys Shared call (line) appearance Enabling call conferencing, call forward, call transfer, and so on. Select ring tones, audio input, and extension mobility settings
Customize individual extension parameters, including:
Ext tab
(1-6, depending on phone model) See Chapter 2, Configuring Lines and Extensions.
Shared line/call appearance NAT settings SIP settings such as SIP debug and SIP port Mailbox ID, MOH server Voice mail server Proxy and registration information Subscriber information such as user ID and password Audio settings Dial plan settings
27
Getting Started
1
In the configuration utility, click the ... User tab See Chapter 3, Customizing Cisco SPA and Wireless IP Phones.
To perform these tasks... Customize user-level parameters, including:
Call forward Speed dial Supplementary services Web information (RSS newsfeeds) Traffic information settings Audio volume Phone GUI settings
View and change parameters for Unit 1 and Unit 2 (applicable only to Cisco SPA300 Series or Cisco SPA500 Series IP Phones with one or two Cisco SPA500S attendant consoles attached)
Attendant Console tab See Chapter 9, Configuring the Cisco SPA500S Attendant Console.
28
Getting Started
Viewing Phone Information
1
After you log on to the configuration utility (see Using the Web-Based Configuration Utility section on page 22), you can check the current status of the Cisco SPA IP Phone by clicking the Info tab. The Info tab shows information about all phone extensions, including phone statistics and the registration status. All fields are read-only. See Info Tab section on page 224 for more information about the fields.
Viewing Phone Information
Using IVR on the Cisco SPA301 and Cisco SPA501G IP Phone

The Cisco SPA301 and Cisco SPA501G provides an IVR menu to perform configuration tasks and obtain information about the phone. To access the IVR menu: Cisco SPA301: Press the asterisk (*) button four times. Enter the number of the option and press #. Some menus require entering of further information or numbers. Cisco SPA501G: Press the Settings button. Enter the number of the option and press #. Some menus require entering of further information or numbers.
You can either press 9 for help, or directly enter the number of the menu option you want. Pressing 9 helps you through a menu of commonly used tasks: Enter the number of the settings you want to change: 1Network 1Connection TypeRecites the connection type. Press 1 to change, then enter 0 for DHCP or 1 for static IP. To save, press 1. To review, press 2. To reenter, press 3. To exit, press *. 2IP AddressRecites the IP address of the phone. 3NetmaskRecites the netmask of the phone. 4Gateway AddressRecites the gateway address of the phone. 5MAC AddressRecites the MAC (hardware) address of the phone.
29
Getting Started
6DNSRecites the primary DNS server address.
2Protocol 1Call Control ProtocolRecites the current call control protocol. Press 1 to change, or * to go back. 2Multicast AddressRecites the multicast address. Press 1 to change, or * to go back. 3CDPTells you if CDP is enabled. Press 1 to change, or * to go back. 4SPCP Auto DetectionTells you if SPCP auto detection is enabled. Press 1 to change, or * to go back.
3 Other Options 1Software VersionRecites the software version. 2Primary ExtensionRecites the primary extension. 3RebootReboots the phone. Hang up to exit. 4Factory ResetRestores the phone to the factory default software and settings. Enter 1 to confirm or * to cancel. 5Debug ServerRecites the address of the debug server. Press 1 to change, or * to go back.
The following table lists the IVR options that you can enter directly after accessing the IVR system.
Number
100 110 120 130 140 150 160
Option
Tells you if Dynamic Host Configuration Protocol (DHCP) is enabled. Recites the IP address of the phone. Recites the netmask of the phone. Recites the gateway address. Recites the MAC (hardware) address of the phone. Recites the phone software version. Recites the primary DNS server address.
30
Getting Started
Number
170 180
Option
Recites the HTTP port on which the web server listens. Defaults to 80. Recites the IP multicast address (used by the Cisco SPA 9000 to communicate with the IP phone). Recites the method of call control (SIP or SPCP). Set call controlenter the value for the call control method you want, then press #:
220 221
0: SIP 1: SPCP
To save, press 1; to review, press 2; to re-enter, press 3; to exit, press *. 73738 Restores the phone to the factory default software and settings. Enter 1 to confirm, or * to exit. If you chose to reset, hang up to exit and begin the restore process. 87778 (Cisco SPA501G) Restore the phones user settings to the default. (Clears all user settings such as speed dials.) Enter 1 to confirm, or * to exit. If you chose to reset, hang up to exit and begin the restore process. Reboot the phone. After entering #, hang up to begin rebooting. Set a static IP address. Enter the address (use * to enter the . value), then press #. To save, press 1; to review, press 2; to re-enter, press 3; to exit, press *. NOTE DHCP must be disabled to use this option; if DHCP is not disabled, you receive an error message. 121 Set a netmask. Enter the address (use * to enter the . value), then press #. To save, press 1; to review, press 2; to re-enter, press 3; to exit, press *. NOTE DHCP must be disabled to use this option; if DHCP is not disabled, you receive an error message.
732668 111
31
Getting Started
Number
131
Option
Set a gateway. Enter the address (use * to enter the . value), then press #. To save, press 1; to review, press 2; to re-enter, press 3; to exit, press *. NOTE DHCP must be disabled to use this option; if DHCP is not disabled, you receive an error message.
161
Set the address of the primary Domain Name Server (DNS). Enter the address (use * to enter the . value), then press #. To save, press 1; to review, press 2; to re-enter, press 3; to exit, press *.
181
Set the IP multicast address (used by the Cisco SPA 9000 to communicate with the IP phone). Enter the address (use * to enter the . value), then press #. To save, press 1. To review the value you entered, press 2. To re-enter, press 3. To exit, press *.
7932
Enable or disable the web-based configuration utility. Press 1 to enable or 0 to disable, then press #. To save, press 1; to review, press 2; to re-enter, press 3; to exit, press *.
723646
Enable or disables access to the administrative (admin) login on the configuration utility. Press 1 to enable or 0 to disable, then press #. To save, press 1; to review, press 2; to re-enter, press 3; to exit, press *.
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2
Configuring Lines and Extensions
This chapter contains the following sections: Configuring Lines, page 33 Configuring Extensions, page 43
Configuring Lines
The Cisco SPA IP Phones (also called stations in this document) provide different numbers of lines depending on the phone model. See the Overview of the Phones section on page 12 for more information. Each line corresponds to a phone number (or extension) used for calls. Each line can support two calls. So, for example, a four-line phone can handle eight calls. One call can be active (in conversation) and seven can be on hold.
Shared Line Appearances

Shared Line Appearance (SLA) allows multiple phones to share an extension number and manage a call as a group. At any given time, each station sharing a call appearance can monitor the state of the call appearance. A station can select a shared call appearance to make a call only if the call appearance is not being used by another station. When a call is made to the extension number for the SLA, all sharing stations ring. Any station can answer the call. If the active phone places the call on hold, the call can be resumed from any of the sharing stations by pressing the corresponding line key from another phone (Cisco SPA303 and Cisco SPA500 series, except for the Cisco SPA502G) or the Select button when the Resume icon is displayed (WIP310).
33

Configuring Lines
NOTE The Cisco SPA300 Series and Cisco SPA500 Series IP Phones support the private
hold feature for MetaSwitch and BroadSoft. Users who have a shared line can press the PrivHold softkey, and the call can only be resumed by the user who placed the call on hold. Each station with an SLA can be configured independently. Although the account information is usually the same for all of the stations, settings such as the dial plan or the preferred codec can vary.
Configuring a Line
NOTE This section does not apply to the WIP310.
To configure a phone line:

STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the Phone tab. STEP 4 Under each line key for the phone, configure the following:
ExtensionAssign an extension to the line key. Defaults to 1. Generally you should reserve EXT 1 on the client station as the primary and private extension of the designated user. Configure shared extensions on EXTs 2 through 6 (depending on phone model).
Short NameEnter a short name or number to display on the LCD for the line
key.
Share Call AppearanceSelect shared if you want the line key to share
incoming call appearances with other phones. See Configuring Shared Line Appearance, page 35. If you select private, the call appearance is private and not shared with any other phone. Defaults to private.
Extended FunctionSee Assigning Busy Lamp Field, Call Pickup, and Speed
Dial Functions to Unused Lines on a Cisco SPA300 Series or Cisco SPA500 Series IP Phone, page 36. NOTE The number of line keys displayed depends on the type of phone.
STEP 5 Click Submit All Changes.
34

Configuring Lines
Configuring Shared Line Appearance

After configuring the line and choosing shared in the Shared Call Appearance field, perform the following steps:
STEP 1 Click the Ext <number> tab of the extension that is shared (do not use Ext 1). STEP 2 Under General, in the Line Enable field, choose yes. STEP 3 Under Share Line Appearance, in the Share Ext field, select shared. If you set this
extension to private (not shared), the extension does not share calls, regardless of the Share Call Appearance setting on the Phone tab. If you set this extension to shared, calls follow the Share Call Appearance setting on the Phone tab. On the Cisco SPA50XG phones that have line buttons, a hollow telephone icon is displayed next to the shared line button. For the Cisco SPA525G/525G2, a telephone icon is displayed.
STEP 4 In the Shared User ID field, enter the user ID (name) of the phone with the
extension that is being shared.

STEP 5 (Optional) In the Subscription Expires field, enter the number of seconds before
the SIP subscription expires. Before the subscription expiration, the phone gets NOTIFY messages from the SIP server on the status of the shared phone extension. The default is 60 seconds.
STEP 6 Under Proxy and Registration, in the Proxy field, enter the IP address of the proxy
server (for example, the IP address of the Cisco SPA 9000).

STEP 7 Under Subscriber Information, enter a Display Name and User ID (extension
number) for the shared extension. These are shown on the phone screen.
STEP 8 (Optional) In the Phone tab, under Miscellaneous Line Settings, you can configure
line mapping. Each LED (line/extension) can hold two calls. You can assign an extension to two LEDs. The first call always causes the assigned LED to flash. Choose one of the following: Vertical firstThe next LED on the phone flashes with the second incoming call. Horizontal firstThe same LED to flashes with the second incoming call.
STEP 9 (Optional) Under SCA Barge-In Enable, choose yes to allow users sharing call
appearances to take over the call on a shared line.
35

Configuring Lines
For example, Bob and Chris share the extension 401. A caller, Adam, calls extension 401. Bob answers the call. Adam and Bob are connected. If Chris has the SCA Barge-In Enable field on her phone set to yes, she can press the line button for extension 401. Then Chris and Adam are connected in a call and Bob is dropped from the call.
NOTE The Cisco SPA525G/525G2 supports the private hold feature for MetaSwitch
and Broadsoft. Users who have a shared line can press the PrivHold softkey, and the call can only be resumed by the user who placed the call on hold. No barge-in can be performed on these calls.
Assigning Busy Lamp Field, Call Pickup, and Speed Dial Functions to Unused Lines on a Cisco SPA300 Series or Cisco SPA500 Series IP Phone
You can configure unused or idle lines on a Cisco SPA300 Series or Cisco SPA500 Series IP Phone to interact with another phone line in the system. For example, if you have two idle lines on an assistants phone, you can configure those lines to show the status of a supervisors phone (Busy Lamp Field, or BLF). You can also configure the idle lines so that they can be used to speed dial the supervisors phone, or pick up calls that are ringing on the supervisors phone. A monitored extension must be private, not shared. Additionally, if using the Cisco SPA9000 for call control, an extension can only be monitored by one other extension. TIP For detailed instructions on configuring the phones with the BroadSoft Busy Lamp Field (BLF) feature, see Configuring SPA303 and 5xxG IP Phones with Broadsoft's BLF, available on the Cisco Support Community at: https://supportforums.cisco.com/docs/DOC-9977
Configuring Call Pickup and Busy Lamp Field

NOTE You must enable BLF to configure call pickup.
In this example, the assistant Bob (extension 200) has an idle line (line 4) on his Cisco SPA508G. He would like to be able to see if his supervisor Stephanie (extension 300) is on the phone, and pick up calls that are ringing at her extension.
36

Configuring Lines
To configure this feature for Bobs Cisco SPA508G:

STEP 1 Log in to the web configuration utility. STEP 2 Click Admin Login. STEP 3 Click advanced. STEP 4 Click the Phone tab. STEP 5 Find the first line to configure (line 4 in this example):
a. From the Extension drop down list, choose Disabled. b. From the Share Call Appearance drop-down list, choose private. c. Enter the following string in the Extended Function field: fnc=blf+cp;sub=Stephanie@$PROXY;ext=300@$PROXY Using the following syntax: fnc=type;sub=stationname@$PROXY;ext=extension#@$PROXY where: fnc: function blf: busy lamp field cp: call pickup sub: station name ext or usr: extension or user (the usr and ext keywords are interchangeable)
STEP 6 Click Submit All Changes. After the phone reboots, the phone (in this example)
should show the following color LEDs for the monitored lines: Green: Available Red: Busy Red Fast Blink: Ringing
37

Configuring Lines
If the phone LEDs display orange or slow blinking orange, there is a problem: Orange denotes that the phone failed to subscribe (received 4xx response) and slow blinking orange denotes an undefined problem (there may be no response to subscribe, or BLF).
In this example, after this configuration, Bob will be able to monitor Stephanies line. He can press line button 4 to pick up a call ringing at Stephanies line.
Configuring Speed Dial

In this example, the assistant Bob (extension 200) has another idle line (line 5) on his Cisco SPA508G. He would like to be able to speed dial his supervisor Mark (extension 400) from that line. To configure this feature for Bobs Cisco SPA508G:
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login. STEP 3 Click advanced. STEP 4 Click the Phone tab. STEP 5 Find the first line to configure (line 5 in this example):
a. From the Extension drop down list, choose Disabled. b. From the Share Call Appearance drop-down list, choose private. c. Enter the following string in the Extended Function field: fnc=sd;ext=400@$PROXY Using the following syntax: fnc=type;ext=extension#@$PROXY where: fnc: function sd: speed dial ext or usr: extension or user (the usr and ext keywords are interchangeable)
38

Configuring Lines
In this example, after this configuration, Bob can press line button 5 to dial Marks line.
Configuring Unused Line Keys for Call Park on the Cisco SPA525G/525G2 (MetaSwitch)
You can configure unused line keys for call park (for the MetaSwitch softswitch) on the Cisco SPA525G/525G2. Users can then press this line button to park a call or retrieve a parked call. To configure:
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login. STEP 3 Click advanced. STEP 4 Click the Attendant Console tab. In the General section, under Server Type,
choose RFC3265_4236.
STEP 5 Click the Phone tab. STEP 6 Choose the line key to configure (line 5 in this example):
a. From the Extension drop down list, choose Disabled. b. From the Share Call Appearance drop-down list, choose private. c. Enter the following string in the Extended Function field: fnc=prk;[email protected] where: fnc: function prk: call park sub: call park orbit, or location where the call is parked. Valid value range is from 01 through 10. In this example, 5 is used. domain.com: phone domain, usually the same as the proxy value in the Ext 1 tab. You can also use fnc=prk;sub=05@$PROXY to use this value.
39

Configuring Lines
Configuring Unused Line Keys to Access Services

On the Cisco SPA300 Series or Cisco SPA500 Series IP Phones, unused or idle phone lines can also be configured to access services, such as the following: XML services MP3 player (Cisco SPA525G/525G2) Weather (RSS) News (RSS)
To configure line keys to access services:

STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login. STEP 3 Click advanced. STEP 4 Click the Phone tab. STEP 5 Find the first line to configure (line 4 in this example):
a. From the Extension drop down list, choose Disabled. b. Enter the following string in the Extended Function field: fnc=type where: fnc: function type: choose from the following: xml: pressing the line button accesses XML services.
NOTE The XML service configured on the Phone tab under the XML
Service field is used (see the Configuring XML Services section on page 91). You can specify a different XML service to connect to by using the syntax fnc=xml;URL=http://xxx.xx.xxx/entry.html where xxx.xx.xxx is the URL of the XML service.
40

Configuring Lines
mp3: pressing the line button starts the mp3 player. weather: pressing the line button accesses weather information. news: pressing the line button accesses news.
For example, to configure line 4 for the mp3 player: fnc=mp3

STEP 6 Click Submit All Changes. After the phone reboots, configured lines glow orange
and display the following icons next to the extension label: xml: XML icon mp3: mp3 player icon (Cisco SPA525G/525G2) news: RSS icon weather: thermometer icon
Configuring Line Key LED Patterns on the Cisco SPA300 Series or Cisco SPA500 Series IP Phone
You can customize the LED patterns for the line keys on the phone by entering letters for the color or pattern in the LED pattern fields. To configure Line Key LED patterns:
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the Phone tab. STEP 4 Under Line Key LED Pattern, use the following letters to customize the fields
shown in the following table: p indicates pattern: the blinking pattern of the LED c indicates color: the color of the LED r indicates red: a red-colored LED g stands for green: a green-colored LED
41

Configuring Lines
2
Description
Appears when the line is idle. Defaults to blank (c=r).
Parameters
Idle LED
Remote Undefined LED
LED pattern during the Remote Undefined state, where the shared call state is undefined (the station is still waiting for the state information from the application server). Not applicable if the call appearance is not shared. Leaving this entry blank indicates the default value of c=r;p=d. Appears when this station seizes the call appearance to prepare for a new outbound call. Defaults to blank (c=r).
Local Seized LED
Remote Seized LED (applicable only to shared call appearance) Local Progressing LED
Appears when the shared call appearance is seized by another station. Defaults to blank (c=r; p=d).
Appears when this station attempts an outgoing call on this call appearance (the called number is ringing). Defaults to blank (c=r).
Remote Progressing LED (applicable only to shared call appearance) Local Ringing LED
Appears when another station attempts an outbound call on this shared call appearance. Defaults to blank (c=r; p=d).
Appears when the call appearance is ringing. Defaults to blank (c=r;p=f).
Remote Ringing LED (applicable only to shared call appearance) Local Active LED
Appears when the shared call appearance is in ringing on another station. Defaults to blank (c=r;p=d).
Appears when the call appearance is engaged in an active call. Defaults to blank (c=r).
42

2
Description
Appears when another station is engaged in an active call on this shared call appearance. Defaults to blank (c=r;p=d).
Parameters
Remote Active LED (applicable only to shared call appearance) Local Held LED
Appears when the call appearance is held by this station. Defaults to blank (c=r;p=s).
Remote Held LED (applicable only to shared call appearance) Register Failed LED
Appears when another station places this call appearance on hold. Defaults to blank (c=r,p=s).
LED pattern when the corresponding extension has failed to register with the proxy server. Leaving this entry blank indicates the default value of c=a. LED pattern when the Call Appearance is disabled (not available for any incoming or outgoing call). Leaving this entry blank indicates the default value of c=o. Appears when the corresponding extension tries to register with the proxy server. Defaults to blanks (c=r;p=s).
Disabled LED
Registering LED
Call Back Active LED
Indicates Call Back operation is currently active on this call. Defaults to blank (c=r;p=s).
For more information on LEDs, see the Creating an LED Script section on page 220.
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced.
43

STEP 3 Click the Ext <number> tab for the extension you want to configure. STEP 4 In the General section, make sure that Line Enable is set to yes.
You can configure many parameters differently for different extensions. These parameters are grouped on the Ext <number> tab. These parameters are explained in other sections of this document: NAT, Network, and SIP SettingsChapter 4, Configuring SIP, SPCP, and NAT. Call Feature SettingsChapter 3, Customizing Cisco SPA and Wireless IP Phones. Proxy and RegistrationChapter 4, Configuring SIP, SPCP, and NAT. Subscriber InformationChapter 4, Configuring SIP, SPCP, and NAT. Audio (Codec) ConfigurationChapter 5, Configuring Security, Quality, and Network Features. Dial PlanChapter 3, Customizing Cisco SPA and Wireless IP Phones.
44
3
Customizing Cisco SPA and Wireless IP Phones
This chapter describes customizing the SPA IP phones and contains the following sections: Configuring Phone Information and Display Settings, page 46 Configuring Linksys Key System Parameters, page 54 Enabling Call Features, page 54 Customizing Phone Softkeys, page 62 Configuring the Message Waiting Indicator, page 69 Configuring Ring Tones, page 70 Configuring RSS Newsfeeds on the Cisco SPA525G/525G2 IP Phone, page 74 Configuring Audio Settings, page 75 Enabling Wireless (Cisco SPA525G/525G2 only), page 77 Configuring Bluetooth (Cisco SPA525G/525G2 only), page 77 Enabling SMS Messaging, page 82 Enabling the Web Server, page 84 Configuring Lightweight Directory Access Protocol (LDAP) for the Cisco SPA300 Series and Cisco SPA500 Series IP Phones, page 85 Configuring BroadSoft Settings (Cisco SPA300 Series and Cisco SPA500 Series), page 89 Configuring XML Services, page 91
45

Configuring Music On Hold, page 93 Configuring Extension Mobility, page 94 Configuring Video Surveillance on the Cisco SPA525G/525G2, page 95

The configuration utility allows you to customize the phone and configure settings such as the phone name, background photo, logo, and screen saver.
Configuring the Phone Name

STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the Phone tab. STEP 4 Under General, enter the Station Name, or name for the phone. This name shows
up in the corporate directory.

STEP 5 Click Submit All Changes. The phone reboots.
Configuring Voice Mail

To configure the phone to connect to voice mail:
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the Phone tab. STEP 4 Under General, enter the Voice Mail Number. This is the internal or external phone
number or URL to access the voice mail system. If using an external voice-mail service, the number must include any digits required to dial out and any required area code.
46

STEP 5 (Optional) Enter the Voice Mail Subscribe Interval, or the expiration time, in
seconds, of a subscription to a voice mail server.

Configuring Internal Voice Mail for Each Extension (When Using Cisco SPA 400 for Voice Mail)
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the Ext <number> tab. STEP 4 Under Call Feature Settings, enter the voice mail line number and phone extension
in the Mailbox ID field. For example, 2101 indicates that the Cisco SPA 400 voice mail server is configured on the Cisco SPA 9000 Line 2, phone extension 101.
STEP 5 Enter the IP address of the voice mail server. STEP 6 Click Submit All Changes.
Customizing the Startup Screen

You can create a text or 128 x 48 pixel by 1-bit deep image logo to display when the IP phone boots up. (Not applicable to the WIP310.)
NOTE A logo only displays for a short period after the Cisco logo during the boot
sequence. Cisco SPA303 and Cisco SPA50XG:

47

STEP 3 Click the Phone tab. STEP 4 To display a text logo, in the Text Logo field, enter logo text as follows:
Up to two lines of text Each line must be fewer than 32 characters Insert a new line character (\n) and escape code (%0a) between lines For example, Super\n%0aTelecom will display:
Super Telecom Use the + character to add spaces for formatting. For example, you can add multiple + characters before and after the text to center it.
To display a picture logo: a. In the BMP Picture Download URL field, enter the path as follows: http://192.168.2.244/pictures/image04_128x48.bmp (you can also use TFTP) b. Change Select Logo to BMP Picture.
STEP 5 Click Submit All Changes. The phone reboots, retrieves the .bmp file, and displays
it when it next boots.
Cisco SPA525G/525G2:
STEP 1 Log in to the configuration utility. STEP 2 Admin Login and advanced. STEP 3 Click the User tab. In the Screen section, Text Logo field, enter logo text as follows:
Up to two lines of text Each line must be fewer than 32 characters Insert a new line character (\n) and escape code (%0a) between lines For example, Super\n%0aTelecom will display:
Super Telecom
48

Use the + character to add spaces for formatting. For example, you can add multiple + characters before and after the text to center it.
STEP 4 In the Logo Type field, select Text Logo. STEP 5 Click Submit All Changes. The phone reboots.
Changing the Display Background (Cisco SPA300 Series and Cisco SPA500 Series)
You can use a picture to customize the background on your IP phone LCD displays. Phone models and acceptable image file types are: Cisco SPA303 and Cisco SPA50XG: Bitmap format, 1 bit-per-pixel color, size 128 x 48 pixels. Cisco SPA525G/525G2: Either .jpg format (recommended) or bitmap (1, 2, 4, 8, or 24 bits-per-pixel). Recommended image size is 320 x 240 pixels. Other image sizes are scaled to fit, which can cause distortion.
NOTE The phone does not reboot after you change the background image URL.
Cisco SPA303 and Cisco SPA50XG:

NOTE A background image is displayed while the phone is running; a logo is only
displayed during the phone's boot sequence.

STEP 1 Copy the image to a TFTP or HTTP server that is accessible from the phone. STEP 2 Log in to the configuration utility. STEP 3 Click Admin Login and advanced. STEP 4 Click the Phone tab. STEP 5 In the Select Background Picture field, select BMP Picture. STEP 6 Enter the URL of the image file you want in the BMP Picture Download URL field.
The URL must include the TFTP/HTTP server name (or IP address), directory, and filename, for example: tftp://myserver.mydomain.com/images/downloadablepicture.bmp or http://myserver.mydomain.com/images/downloadablepicture.bmp
49

If the HTTP Refresh Timer is set in the servers response to BMP Picture Download URL, the phone downloads the picture from the link and displays it on the screen. The phone automatically retrieves the picture after the specified number of seconds.
When the BMP Picture Download URL is changed, the phone compares the URL to the previous images URL. (If the URLs are the same, the phone does not perform the download.) If the URLs are different, the phone downloads the new image and displays it (providing the Select Background Picture field is set to BMP Picture). Cisco SPA525G:
STEP 1 Copy the image to an HTTP server that is accessible from the phone. (TFTP is not
supported.)
STEP 2 Log in to the configuration utility. STEP 3 Click Admin Login and advanced. STEP 4 Click the User tab. STEP 5 In the Screen section, Background Picture Type field, select Download BMP
Picture.
STEP 6 Enter the URL of the .bmp file you want in the BMP Picture Download URL field.
The URL must include the HTTP server name (or IP address), directory, and filename, for example: http://myserver.mydomain.com/images/downloadablepicture.jpg If the HTTP Refresh Timer is set in the servers response to BMP Picture Download URL, the phone downloads the picture from the link and displays it on the screen. The phone automatically retrieves the picture after the specified number of seconds.
When the BMP Picture Download URL is changed, the phone compares the URL to the previous images URL. (If the URLs are the same, the phone does not perform the download.) If the URLs are different, the phone downloads the new image and displays it (providing the Select Background Picture field is set to Download BMP Picture).
50

Configuring the Screen Saver

You can configure a screen saver for the Cisco SPA300 Series and Cisco SPA500 Series IP Phones. (Not applicable to WIP310.) This option enables a screen saver on the phones LCD. When the phone is idle for a specified time, it enters screen saver mode. (Users can set up screen savers directly using phones Setup button.) Any button press or on/off hook event triggers the phone to return to its normal mode. If a user password is set, the user must enter it to exit screen saver mode. To configure the screen saver: Cisco SPA303 or Cisco SPA50XG:
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the Phone tab. STEP 4 In the General section, in the Screen Saver Enable field, choose yes. STEP 5 In the Screen Saver Wait field, enter the number of seconds of idle time to elapse
before the screen saver starts.

STEP 6 In the Screen Saver Icon field, choose the display type:
A background picture. The station time in the middle of the screen. A moving padlock icon. When the phone is locked, the status line displays a scrolling message Press any key to unlock your phone. A moving phone icon. The station date and time in the middle of the screen. A blank power save screen.
51

STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the User tab. STEP 4 Under Screen, in the Screen Saver Enable field, choose yes. STEP 5 In the Screen Saver Type field, choose the display type:
Black BackgroundDisplays a black screen. Gray BackgroundDisplays a gray screen. Black/Gray RotationThe screen incrementally cycles from black to gray. Picture RotationThe screen rotates through available pictures on the phone. Digital FrameShows the background picture.
STEP 6 In the Screen Saver Trigger Time field, enter the number of seconds that the phone
remains idle before the screen saver turns on.

STEP 7 In the Screen Saver Refresh Time field, enter the number of seconds before the
screen saver should refresh (if, for example, you chose a rotation of pictures).
Configuring the LCD Contrast

You can configure the LCD contrast on the Cisco SPA300 Series and Cisco SPA500 Series IP Phone. (Not applicable to the WIP310.) To configure the contrast for the LCD screen on the phone: Cisco SPA303 and Cisco SPA50XG:
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the User tab.
52

STEP 4 Under LCD, in the LCD Contrast field, enter a number value from 1 to 30. The higher
the number, the greater the contrast on the screen.

STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the User tab. STEP 4 Under Screen, in the LCD Contrast field, enter a number value from 1 to 30. The
higher the number, the greater the contrast on the screen.

Configuring Back Light Settings (Cisco SPA525G/525G2)

To configure the back light settings for the LCD screen on the phone:
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the User tab. STEP 4 Under Screen, in the Back Light Enable field, choose yes to enable the screen
back light.
STEP 5 In the Back Light Timer field, enter the number of seconds of idle time that can
elapse before the back light turns off.

53

Configuring Linksys Key System Parameters
Configuring Linksys Key System Parameters

To configure the phone as part of a Linksys Key System (for use with the Cisco SPA 9000):
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the SIP tab. STEP 4 Under Linksys Key System Parameters, configure the following fields:
Linksys Key SystemEnables or disables the Linksys Key System for use with the Cisco SPA 9000. Defaults to yes. See the Cisco SPA 9000 System Administration Guide for more details. Multicast AddressUsed by the Cisco SPA 9000 to communicate with Cisco IP phones. Defaults to 224.168.168.168:6061. (For the Cisco SPA501G, can be configured using the IVR. See the Using IVR on the Cisco SPA301 and Cisco SPA501G IP Phone section on page 29.) Key System Auto DiscoveryEnables or disables auto discovery of the call control server (for example, the Cisco SPA 9000). Disable this feature for teleworkers or other scenarios where multicast does not work. Key System IP AddressIP address of the call control server IP. Enter the IP address for teleworkers or other scenarios where multicast does not work. Force LAN CodecUsed with the Cisco SPA 9000. Choices are none, G.711u, or G.711a. Defaults to none.
Enabling Call Features

This section describes how to enable and disable call features on the phone.
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Enabling Anonymous Call and Caller ID Blocking Services

STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the Phone tab. STEP 4 Under Supplementary Services, under the type of call blocking to enable, choose
yes. Block ANC ServBlocks anonymous calls. Block CID ServBlocks outbound caller ID.
NOTE These features can also be configured from the User tab, under Supplementary
Services.
Enabling Automatic Call Distribution (ACD) Service

Typically used for call centers, Automatic Call Distribution (ACD) handles incoming calls and manages them based on a database of instructions. You can enable this with the SIP B parameter (Configuring SIP section on page 101). Defaults to no (disabled).
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the Phone tab. STEP 4 Under Supplementary Services, in the ACD Login Serv field, choose yes. STEP 5 In the ACD Ext field, choose the extension used for handling ACD calls. Select 1-6,
depending on your phone model. Defaults to 1.

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Enabling Call Back Service

Call back is a feature that forces the phone to repeatedly try a number that has been dialed and received a busy response. The busy number is tried until the call goes through and the phone rings on the users end. To enable call back service:
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the Phone tab. STEP 4 In the Phone tab, under Supplementary Services, in the Call Back Serv field,
choose yes.
Enabling Call Park and Call Pickup Services

Call park and call pickup are features available on IP phones in a Cisco SPA 9000 system. Call park allows users to put a call on a line and make it available for another user to pick up. Call pickup allows a user to pick up a phone that is ringing at another users phone.
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the Phone tab. STEP 4 Under Supplementary Services, under the type of call feature to enable, choose
yes. Call Park ServEnables call parking. Call Pickup ServEnables call pickup.
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Enabling Call Transfer and Call Forwarding Services

STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the Phone tab. STEP 4 Under Supplementary Services, under the transfer type you want to enable,
choose yes: Attn Transfer ServAttended call transfer service. The user answers the call before transferring it. Blind Transfer ServBlind call transfer service. The user transfers the call without speaking to the caller.
You can also enable or disable call forwarding: Cfwd AllForwards all calls. Cfwd BusyForwards calls only if the line is busy. Cfwd No AnsForwards calls only if the line is not answered.
Enabling Conferencing
To allow the user to perform call conferencing on the phone:
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the Phone tab. STEP 4 Under Supplementary Services, in the Conference Serv field, choose yes. STEP 5 Click Submit All Changes.
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Enabling Do Not Disturb

You can allow users to turn the Do Not Disturb feature on and off. This feature directs all incoming calls to voice mail or, if voice mail is not configured, plays a message to the caller saying the user is unavailable.
NOTE On the Cisco SPA300 Series and Cisco SPA500 Series IP Phones, users can press
the Ignore softkey to divert a ringing call to the forwarded destination. To allow users to use Do Not Disturb (this is enabled by default):
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the Phone tab. STEP 4 Under Supplementary Services, under DND Serv, choose yes. NOTE This feature can also be configured from the User tab, under Supplementary
Services.
Enabling the Missed Call Shortcut

The IP phones can display a notification that a call has been missed. (Not applicable to WIP310.) To enable this notification:
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the User tab. STEP 4 Under Supplementary Services, in the Miss Call Shortcut field, choose yes. STEP 5 Click Submit All Changes.
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Logging Missed Calls (Cisco SPA300 Series and Cisco SPA500 Series)
You may want to disable or enable missed call logging per extension. For example, if you have set up a line to monitor another users line, you may want to disable missed call logging for the monitored line.
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the User tab. STEP 4 Under Supplementary Services, in the Log Missed Calls for EXT <number> field,
choose yes.
Enabling Paging (Intercom)

The paging, or intercom feature, allows two types of paging:
Single Page
A user can directly contact another user by phone. If the person being paged has configured their phone to automatically accept pages (see Configuring a Phone to Automatically Accept Pages, page 60), the phone does not ring and a direct connection between the two phones is automatically established when paging is initiated.
Group Paging
Group Paging lets the user page all the client stations at once, or page groups of phones. If the client station is on an active call while a group page starts, the incoming page is ignored. When paging occurs, the speaker on the paged stations is automatically powered on unless the handset or headset is being used. Group page is one-way only. The paged client stations can only listen to the call from the originator.
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To enable paging:
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the Phone tab. STEP 4 Under Supplementary Services, under Paging Serv, choose yes. STEP 5 Click Submit All Changes.
Configuring a Phone to Automatically Accept Pages

To configure a phone to automatically accept pages:
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the User tab. STEP 4 Under Supplementary Services, in the Auto Answer Page field, choose yes. STEP 5 Click Submit All Changes. The phone reboots.
Configuring Paging Groups

You can configure a phone as part of a paging group. Users can then direct pages to specific groups of phones. Limitations: A phone can be a listening member of no more than two paging groups. No more than five paging groups can be configured on a phone.
To configure a phone as part of a paging group:

STEP 1 Log in to the configuration utility for the phone. STEP 2 Click Admin Login and advanced. STEP 3 Click the Phone tab.
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STEP 4 Under Multiple Paging Group Parameters, enter the paging commands into the
Group Paging Script field. The syntax is as follows:

pggrp=ip-address:port;[name=xxx;]num=xxx;[listen={yes|no}]];
Where: IP address: Multicast IP address of the phone that will listen for and receive pages. port: Port on which to page; you must use different ports for each paging group. All phones in the same paging group must use the same port number. name (optional): The name of the paging group. num: The number users will dial to access the paging group; must be unique to the group. listen: If the phone being configured is a listening member of the page group. A phone can be a listening member of a maximum of two groups. If no value is entered, the default is to not listen as a member of this group.
Configuration Example The following example sets up four paging groups: All, Sales, Support , and Engineering. Users will press 801 to send pages to all phones, 802 to send pages to phones configured as part of the Sales group, 803 to send pages to phones configured as part of the Support group, and 804 to send pages to phones configured as part of the Engineering group. A phone that is configured with this example is a listening member of the All and Sales paging groups. That phone will automatically receive pages sent to those two paging groups. For each Sales phone, enter the following in the Phone > Multiple Paging Groups Parameters > Group Paging Script field:
pggrp=224.123.123.121:43210;name=All; num=801;listen=yes; pggrp=224.123.123.121:43211;name=Sales;num=802; listen=yes; pggrp=224.123.123.121:43212;name=Support;num=803; pggrp=224.123.123.121:43213;name=Engineering;num=804;
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Enabling Secure Call

See Setting Security Features, page 128.
Enabling Service Announcements

The Service Announcements features allows a user to send announcement requests to a customer-supplied announcement server. (Not applicable to the WIP310.)
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the Phone tab. STEP 4 Under Supplementary Services, in the Service Annc Serv field, choose yes. STEP 5 Click Submit All Changes.

NOTE This feature is unavailable on the Cisco SPA300 Series and Cisco SPA500 Series
IP Phones using SPCP. The Cisco SPA300 Series and Cisco SPA500 Series IP phones have four softkeys on the screen that, when pressed, perform certain actions. (The Cisco SPA301 and Cisco SPA501 do not have any softkeys.) The default softkeys (when the phone is in an idle state) are Redial, Directory, Call Forward, and Do Not Disturb. Other softkeys are available during specific call states (for example, if a call is on hold, the Resume softkey displays). You can customize the softkeys displayed on the phone. To program softkeys:
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the Phone tab. STEP 4 (Cisco SPA525G/525G2 only) Under Programmable Softkey Enable, choose yes.
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STEP 5 Edit the softkeys depending on the call state in which you want the softkey to
display. See the following table for information about softkeys.

In the Programmable Softkeys section, each phone state is displayed and the softkeys that are available to display during that state are listed. Each softkey is separated by a semicolon. Softkeys are shown in the format:
softkeyname|[position]
where softkeyname is the name of the key and position is where the key is displayed on the phone screen. Positions are numbered, with position one displayed on the lower left of the screen, followed by positions two through four. Additional positions are accessed by pressing the right arrow key on the phone. If no position is given for a softkey, the key will float and appears in the first available empty position on the screen.
NOTE On the Cisco SPA525G/SPA525G2, in the Off Hook State, the More softkey is fixed
in position 4 and cannot be changed. The table below lists each softkey and the phone state under which the softkey displays. You can have a maximum of 16 softkeys for each call state field.
Keyword
acd_login
Key Label
Login
Definition
Logs user in to Automatic Call Distribution (ACD). Logs user out of ACD. Enter alphabetic characters in a data entry field. Answers an incoming call. Denotes that a user who is logged in to an ACD server has set his status as available. Allows another user to interrupt a shared call.
Available Phone States

Idle
acd_logout alpha
Logout Alpha
Idle Off-Hook, Dialing Input Ringing Idle
answer avail
Answer Avail
barge
Barge
Shared-Active, Shared-Held
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3
Definition
Performs a blind call transfer (transfers a call without speaking to the party to whom the call is transferred). Requires that Blind Xfer Serv is enabled. Cancels a call (for example, when conferencing a call and the second party is not answering). Forwards all calls to a specified number.
Keyword
bxfer
Key Label
BlindXfer/ bxfer

Connected, Connected
cancel
Cancel
Dialing Input
cfwd
Forward
Idle, Off-Hook, Hold, SharedActive, SharedHeld Idle
chkcfwd
Clr Fwd/cfwd Clr DND/dnd Clear Conf
Deactivates call forwarding.
chkdnd
Deactivates Do Not Disturb.
Idle
clear conf
Clears an entire text/number field. Initiates a conference call. Requires that Conf Serv is enabled and there are two or more calls that are active or on hold. Conferences active lines on the phone. Requires that Conf Serv is enabled and there are two or more calls that are active or on hold. Deletes a character when entering text. Dials a number. Provides access to phone directories.
Input Connected, StartConf
confLx
Conf Line
Connected
delchar
delChar
Dialing (input)
dial dir
Dial Dir
Dialing (input) Idle, Connected, Start-Conf, StartXfer, Off-Hook (no input), Redial
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3
Definition
Sets Do Not Disturb to prevent calls from ringing the phone.
Keyword
dnd
Key Label
DND

Idle, Off-Hook (no input), Hold, Shared-Active, Shared-Held Idle Idle
em_login em_logout
Login Logout
Logs user in to Extension Mobility. Logs user out of Extension Mobility. Ends a call.
endcall
End Call
Connected, Offhook, Progressing, Start-Xfer, StartConf, Conferencing, Releasing, Resume Idle, Off-Hook (no input)
gpickup
GrPickup/ grPick
Allows user to answer a call ringing on an extension by discovering the number of the ringing extension. Put a call on hold.
hold
Hold
Connected, StartXfer, Start-Conf, Conferencing Ringing Conferencing Idle, Missed-Call, Off-Hook (no input) Dialing Input Missed-Call Hold, SharedActive Off-Hook (no input), Dialing (input)
ignore join lcr
Ignore Join Call Rtn/lcr
Ignores an incoming call. Connects a conference call. Returns the last missed call.
left miss newcall
Left Miss New Call
Moves the cursor to the left. Displays the list of missed calls. Begins a new call.
option
Option
Opens a menu of input options.
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3
Definition
Puts a call on hold at a designated park number. Puts a call on hold on an active shared line. Allows user to answer a call ringing on another extension by entering the extension number. Displays the redial list.
Keyword
park
Key Label
Park

Connected
phold
PrivHold
Connected
pickup
Pickup
Idle, Off-Hook (no input)
redial
Redial
Idle, Connected, Start-Conf, StartXfer, Off-Hook (no input), Hold Idle, Hold, SharedHeld Dialing (input) Off-Hook, Dialing (input)
resume
Resume
Resumes a call that is on hold.
right starcode
Right Input Star Code/ *code Toggle
Moves the cursor to the right. Displays a list of star codes that can be selected.
toggle
Switches between two calls that are active or on hold. (Cisco SPA502) Denotes that a user who is logged in to an ACD server has set his status as unavailable. Resumes a parked call.
Connected
unavail
Unavail
Idle
unpark
Unpark
Idle, Off-Hook (no input) Connected, StartXfer
xfer
Transfer/ xfer
Performs a call transfer. Requires that Attn Xfer Serv is enabled and there is at least one connected call and one idle call. Transfers an active line on the phone to a called number. Requires that Attn Xfer Serv is enabled and there are two or more calls that are active or on hold.
xferLx
Xfer Line/ xferLx
Connected
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Programmable Softkeys
The Cisco SPA300 Series and Cisco SPA500 Series IP Phones provide six programmable softkeys (fields PSK 1 through PSK 6). These keys can be defined by either a speed dial script or an XML service script. To configure programmable softkeys:
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the Phone tab. STEP 4 (Cisco SPA525G/525G2 only) Under Programmable Softkey Enable, choose yes.
To configure a speed dial script, enter the following in the PSK field: fnc=sd;ext=extensionname@$PROXY;vid=outboundextnum;nme=name where fnc is the function of the key (speed dial), ext (extensionname) is the extension being dialed, vid is the extension on the calling phone from which the outbound call is sent, and name is the name of the speed dial being configured. Vertical service activation codes (* codes) are supported in speed dial configurations.
NOTE The name field displays on the softkey on the phone display screen. Cisco
recommends a maximum of 8 characters for a Cisco SPA30X or Cisco SPA50X phone and 10 characters for a Cisco SPA525G/525G2 phone. If more characters are used, the label can be truncated on the phone display. To configure an XML script, enter the following in the PSK field: fnc=xml;url=http://scriptURL.xml;nme=scriptname where fnc is the function of the key (an XML script), scriptURL.xml is the URL where the script is located, and scriptname is the name of the script.
NOTE The scriptname field displays on the softkey on the phone display screen. Cisco
recommends a maximum of 8 characters for a Cisco SPA30X or Cisco 50X phone and 10 characters for a Cisco SPA525G/525G2 phone. If more characters are used, the label can be truncated on the phone display.
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You can use macro variables in XML URLs. The following macro variables are supported: User IDUID1, UID2 Display nameDISPLAYNAME1, DISPLAYNAME2 Auth IDAUTHID1, AUTHID2 ProxyPROXY1, PROXY2 MAC AddressMA Product NamePN Product Series NumberPSN Serial NumberSERIAL_NUMBER
Configuration Example
You want to configure the Cisco SPA525G/525G2 phone with softkey that, when pressed, dials the Sales Departments extension (200). You want this button to display on the far lower left of the screen when the phone is idle, when the phone is off hook, or when the phone is connected on a call. You want the outbound call (that is going to the speed dial) to originate from the second extension on the users phone, not the primary extension.
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the Phone tab. STEP 4 Under Programmable Softkey Enable, choose yes. STEP 5 In the Programmable Softkeys section, edit the following:
Programmable Softkey Enable: yes PSK1: fnc=sd;ext=200@$PROXY;vid=2;nme=Sales
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Configuring the Message Waiting Indicator
Idle Key List: Edit the field to add psk1|1 to the beginning of the string; for example: psk1|1;em_login;acd_login;acd_logout;avail;unavail; redial;dir;cfwd;dnd;lcr;pickup;gpickup;unpark;em_logout;
Off Hook Key List: Edit the field to add psk1|1 to the beginning of the string; for example: psk1|1;option;redial;dir;cfwd;dnd;lcr;unpark;pickup; gpickup;
Connected Key List: Edit the field to add psk1|1 to the string, editing the existing softkeyname|1 to PSK1. For example, the original string: hold|1;endcall|2;conf|3;xfer|4;bxfer;confLx;xferLx;park;p hold;flash; becomes: psk|1;hold|2;endcall|3;conf|4;xfer;bxfer;confLx;xferLx; park;phold;flash
STEP 6 Click Submit All Changes. The Sales speed dial softkey is displayed in the lower
left of the screen when the phone is idle, when the phone is connected on a call, and when the phone is off hook.
Configuring the Message Waiting Indicator

You can configure the message waiting indicator (MWI) for separate extensions on the phone. The MWI lights based on the presence of new voicemail messages in the mailbox. However, if the indicator at the top of your Cisco SPA300 Series or Cisco SPA500 Series IP Phone is not lighting when voice mail is left, or you are not seeing message waiting notifications on your WIP310:
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the Ext <number> tab. STEP 4 Under Call Feature Settings, in the Message Waiting field, choose yes.
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Configuring Ring Tones

You can define up to ten ring tones for a Cisco SPA300 Series or Cisco SPA500 Series IP Phone.
NOTE WIP310 ring tones are not configurable from the configuration utility.
You can define: The default ring tone for the extension Specific ring tones assigned to individual callers in the personal directory. These override the default ring tone.
To configure ring tones:

STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the Phone tab and proceed to the Ring Tone section.
You can configure the characteristics of each ring tone using a Ring Tone script. In a Ring Tone script, you can assign a name for the ring tone, and specify: Name (n)Ring tone name, such as Classic, Simple, and Office Waveform (w)1, 2, 3, or 4 Cadence (c)1, 2, 3, 4, or 5
You can also download one of two available ring tones (user ring tone 1 or 2) using TFTP: http://phone_ip_addr/ringtone1?[url] Where the URL syntax is tftp://host[:port]/path. The default host is the TFTP host. Port is optional. The default port is 69. The link is case sensitive.
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On the IP phones, user-downloaded ring tones are labeled User 1 and User 2 in the choices for the Default Ring. On the phone ring tone menu, the User 1 and 2 choices are replaced by the corresponding name of the ring tone. Not Installed appears if the user ring tone slots are not used. For ring tone User 1 and User2, the cadence is fixed with the on-time equals to the duration of the ring tone file and off-time equals to four seconds. The total ring duration is fixed at 60 seconds. The user ring tone names displayed on the phone LCD are derived from the ring tone file header file. The phone does not require rebooting after downloading a ring tone. To remove the User 1 ring tone from the phone, set the path to delete, as follows: http://phone_ip_addr/ringtone1?/delete
Configuring On-Demand Ring Tones (Cisco SPA525G/ 525G2)

The Cisco SPA525G/525G2 supports on-demand ring tones, which means that ring tones are downloaded and played from a TFTP server when a call comes in. To configure:
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the Phone tab. STEP 4 Under Ring Tone, in one or more of the ten ring tone fields, enter the following:
n=office;w=[tftp://]host[:port]/path;c=0
and specify the URL to download in the host/port/path field. If the connection cannot be established, a default ring tone is played.
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User-Created MP3 Ring Tones (Cisco SPA525G/525G2)

Cisco SPA525G/525G2 users can create up to two ring tones from an MP3 audio file stored on a USB memory device. For instructions, see the Cisco Small Business SPA525G/525G2 User Guide (SIP), located on Cisco.com. (See Appendix C, Where to Go From Here, for the location of this document.)
Creating and Uploading Ring Tones Using the Ring Tone Utility (Cisco SPA30X and Cisco SPA50XG only)
To convert a file for use as a ring tone, use the Ring Tone Utility, available at: https://www.myciscocommunity.com/docs/DOC-6672 You must have a .wav file less than 8 seconds in length saved to your computer. You can also use a sound editor to create the file with the following restrictions: 16-bit PCM mono 8000 samples per second less than 6000 ms in length
To create a ring tone and upload it to a phone:

STEP 1 Open the Ring Tone Utility. STEP 2 Enter the IP address of the phone. STEP 3 Click Browse and navigate to the directory on your computer where the source
.wav file is stored. Select the wav file and click Open.
STEP 4 Click Load Source File. STEP 5 Enter a name for the ring tone. This name will appear in the display on the phone.
You choose the file name later.

STEP 6 Enter the target. You can have up to two customized ring tones uploaded to the
phone.
STEP 7 (Optional) Click Preview to preview the ring tone. Click Options to change the start
or end positions, or to squeeze or stretch the audio.

STEP 8 Click Upload to Phone to upload the ring tone to the phone. Click OK when the
success status message appears.
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STEP 9 Close the open Ring Tone Utility windows.
To create a ring tone and save it to a file:

STEP 1 Open the Ring Tone Utility. STEP 2 Enter the IP address of the users phone or press Skip to create the ring tone and
save it as a file.
STEP 3 Click Browse and navigate to the directory on your computer where the source
wav file is stored. Select the wav file and click Open.
STEP 4 Click Load. STEP 5 Enter a name for the ring tone. This name will appear in the phone display. You
choose the file name later.

STEP 6 (Optional) Click Preview to preview the ring tone. Click Options to change the start
or end positions, or to squeeze or stretch the audio.

STEP 7 Click Save As to save the file to your computer. Enter the file name and press Save. STEP 8 Close the open Ring Tone Utility windows.
To delete a ring tone from a phone:

STEP 1 Open the Ring Tone Utility. STEP 2 Enter the IP address of the phone. STEP 3 Click the Delete button next to the ring tone you want to delete. STEP 4 Click OK. STEP 5 Close the open Ring Tone Utility windows.
Assigning a Ring Tone to an Extension

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Configuring RSS Newsfeeds on the Cisco SPA525G/525G2 IP Phone
STEP 3 Click the Ext <number> tab. STEP 4 Under Call Feature Settings, in the Default Ring field, choose from the following:
No Ring 1 through 10 User 1 User 2
Configuring RSS Newsfeeds on the Cisco SPA525G/525G2 IP Phone

The Cisco SPA525G/525G2 provides the option to view RSS newsfeeds for news in the categories of local, world, finance, sports, and politics. Newsfeeds that are provided by Yahoo are supported for U.S. customers only. To configure newsfeeds:
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the User tab. STEP 4 Under Web Information Service Settings, you can edit the following fields:
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Configuring Audio Settings
Parameter
RSS Feed URLs 1-5
Description
URLs for Local and World news, Finance, Sports, and Politics. Default values are:

Weather Temperature Unit
1Local News (defaults to URL http://rss.cnn.com/rss/ cnn_us.rss) 2World News (defaults to URL http://newsrss.bbc.co.uk/ rss/newsonline_uk_edition/world/rss.xml) 3Finance News (defaults to URL http:// finance.yahoo.com/rss/topstories) 4Sports News (defaults to URL http:// rss.news.yahoo.com/rss/sports 5Politics News (defaults to URL http:// rss.news.yahoo.com/rss/politics)
Choose which unit to display for weather information (Fahrenheit or Celsius).

You can configure default audio volume settings for the phone. These settings can be modified by the user by pressing the volume control button on the phone, then pressing the Save soft button. (Not applicable to the WIP310.) To configure the audio volume settings:
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the User tab. You can configure the following settings:
Parameter Ringer Volume
Description Sets the volume for the ringer.
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3
Description Sets the volume for the full-duplex speakerphone. Sets the volume for the handset. Sets the volume for the headset. Sets the volume for the Bluetooth device.
NOTE Applies to Cisco SPA525G/525G2 only.
Parameter Speaker Volume Handset Volume Headset Volume Bluetooth Volume
Configuring Audio Input Gain (Cisco SPA300 Series and Cisco SPA500 Series)
You can amplify or deamplify the sound on your phones handset, headset, and speakerphone.
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the Phone tab. STEP 4 Under Audio Input Gain (dB), choose the item to configure.
If you enter a positive value, amplification increases (sound is louder). If you enter a negative value, amplification decreases (sound is softer). You can enter a value from -6 decibels to +6 decibels. All fields default to zero. Try a value that is loud enough without producing echo (an issue if the input gain is too high).
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Enabling Wireless (Cisco SPA525G/525G2 only)
Enabling Wireless (Cisco SPA525G/525G2 only)

The Cisco SPA525G/525G2 provides a built-in Wireless-G interface. To enable wireless:
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the System tab. STEP 4 Under Wi-Fi Settings, in the SPA525-wifi-on field, choose yes. STEP 5 Click Submit All Changes.
Configuring Bluetooth (Cisco SPA525G/525G2 only)

The Cisco SPA525G/525G2 supports Bluetooth to allow use of the phone with a wireless Bluetooth-enabled headset. The Cisco SPA525 (G2 model only) also supports the use of your Cisco IP phone in conjunction with Bluetooth-enabled mobile phone. You can do the following: Pair your Bluetooth-enabled mobile phone and Cisco SPA525G2. Your mobile phone is assigned a line button on your Cisco SPA525G2. You can make and receive mobile calls using the Cisco SPA525G2. Switch audio for in-progress calls between your mobile phone and the Cisco SPA525G2. Import your mobile phone address book to your Cisco SPA525G2 personal address book. Use your Cisco SPA525G2 as a handsfree device for your mobile phone.
Enabling Bluetooth (Cisco SPA525G/525G2)

To enable Bluetooth from the configuration utility:
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STEP 3 Click the System tab. STEP 4 Under Bluetooth Settings, in the Enable BT field, choose yes. STEP 5 Click Submit All Changes.
To enable Bluetooth from the phone interface:

STEP 1 Press the Setup button. STEP 2 Scroll to User Preferences and press Select. STEP 3 Scroll to Bluetooth Configuration and press Select. STEP 4 With Bluetooth selected, press the Right Arrow key until a blue check mark
appears.
STEP 5 Press Save.
Using a Bluetooth Headset (Cisco SPA525G/525G2)

To enable a Bluetooth headset from the configuration utility:
STEP 1 Log in to the configuration utility. STEP 2 Click Bluetooth. STEP 3 Under Bluetooth Device, choose On. STEP 4 Under Bluetooth Device List, press Scan for Bluetooth Devices. STEP 5 In the Bluetooth Device List, click the name of the Bluetooth headset. STEP 6 Enter the PIN for the Bluetooth headset. STEP 7 Press Submit All Changes.
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To enable a Bluetooth headset from the phone:

STEP 1 Enable Bluetooth as described in the Enabling Bluetooth (Cisco SPA525G/
525G2) section on page 77.

STEP 2 Press the Setup button. STEP 3 Scroll to User Preferences and press Select. STEP 4 Scroll to Bluetooth Configuration and press Select. STEP 5 Scroll to Bluetooth Mode and press the Right Arrow key to choose one of the
following: PhoneYour Cisco SPA525G2 will pair with a Bluetooth headset only. Choose this option if you will not use the Cisco SPA525G2 with a Bluetoothenabled mobile phone. BothYour Cisco SPA525G2 will use a Bluetooth headset, or operate with your Bluetooth-enabled mobile phone (see Pairing Your Cisco SPA525G2 with a Bluetooth-Enabled Mobile Phone, page 80). Note that your Cisco SPA525G2 will connect to only one device at a time (either the Bluetooth headset or the Bluetooth-enabled mobile phone.) If multiple Bluetooth devices are in range of the Cisco SPA525G2, the order of devices in the Bluetooth Configuration > Bluetooth Profiles list is used, and the device with a higher priority is activated first.
STEP 6 Scroll to Bluetooth Profiles and press the Right Arrow key to enter the profile
screen.
STEP 7 Press Scan to scan for your headset. NOTE Depending on the network environment (for example, the number of Bluetooth
devices and noise level), your Bluetooth headset may not appear on the found devices list. Ensure the headset is powered on and has Bluetooth activated, and retry the scan.
STEP 8 In the list of found devices, select your headset and press the Select button to edit
the profile.
STEP 9 Scroll to PIN and enter the PIN for your Bluetooth headset.
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STEP 10 Scroll to Connect Automatically and press the Right Arrow key to turn to On. STEP 11 Press Connect. The profile screen displays and a check mark appears next to the
headset if the connection was successful.
Pairing Your Cisco SPA525G2 with a Bluetooth-Enabled Mobile Phone

This feature is based on the following standard Bluetooth profiles: Phone Book Access Profile 1.0 Handsfree Profile 1.5 Handset Profile 1.1
Make sure your mobile phone provides support for the above profiles. Cisco provides a reference list of Bluetooth-enabled mobile phones supported with the Cisco SPA525G2. See the Cisco support community at http://www.cisco.com/go/ smallbizsupport and also consult the latest Cisco SPA525G2 release notes, available at cisco.com.
NOTE For more detailed instructions, including screen shots, see the Cisco Small
Business SPA525/525G2 User Guide (SIP) or the Cisco Unified Communications Manager Express for the Cisco Small Business IP Phone SPA525G/525G2 (SPCP) User Guide.
To pair your Cisco SPA525G2 with your Bluetooth-enabled mobile phone, you can either initiate pairing from the Cisco SPA525G2, or from your mobile phone.
Initiating Pairing from the Cisco SPA525G2

STEP 1 Enable Bluetooth as described in the Enabling Bluetooth (Cisco SPA525G/
525G2) section on page 77.

STEP 2 Press the Setup button. STEP 3 Scroll to User Preferences and press Select. STEP 4 Scroll to Bluetooth Configuration and press Select.
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STEP 5 Scroll to Bluetooth Mode and press the Right Arrow key to choose one of the
following: HandsfreeYour Cisco SPA525G2 will operate as a handsfree device with a Bluetooth-enabled mobile phone. BothYour Cisco SPA525G2 will operate with your Bluetooth-enabled mobile phone or operate with a Bluetooth headset. Note that your Cisco SPA525G2 will connect to only one device at a time (either the Bluetooth headset or the Bluetooth-enabled mobile phone.) If multiple Bluetooth devices are in range of the Cisco SPA525G2, the order of devices in the Bluetooth Configuration > Bluetooth Profiles list is used, and the device with a higher priority is activated first.
STEP 6 Scroll to Bluetooth Profiles and press the Right Arrow key to enter the profile
screen.
STEP 7 Press Scan to scan for your mobile phone. NOTE Depending on the network environment (for example, the number of Bluetooth
devices and noise level), your Bluetooth headset may not appear on the found devices list. Ensure the headset is powered on and has Bluetooth activated, and retry the Scan.
STEP 8 In the Select a Bluetooth Device to Pair list, select the mobile phone to which you
want to pair and press Connect.
Initiating Pairing from Your Bluetooth-Enabled Mobile Phone

NOTE The procedure varies depending on your phone model. The following example
uses an Apple iPhone. Before starting, its helpful to find the MAC address of your Cisco SPA525G2 IP phone. From your IP phone, go to the Setup menu and select Status. Select Product Information. The MAC address is displayed.
STEP 1 On your iPhone, click Settings. STEP 2 Under General, choose Bluetooth. Ensure Bluetooth is turned on. STEP 3 In the Bluetooth Window, under Devices, find the MAC address of your Cisco
SPA525G2 IP phone.
STEP 4 Select the MAC address of the Cisco SPA525G2.
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Enabling SMS Messaging
STEP 5 Enter the PIN (the default is 0000) and press Connect.
When paired with your mobile phone, the Cisco SPA525G2 display screen assigns one of your line buttons to the mobile phone. A mobile phone icon with a flashing lightning bolt icon is displayed next to the mobile phone number.
To verify the mobile phone configuration:

STEP 1 On the Cisco SPA525G2, press the Setup Button. STEP 2 Scroll to User Preferences and press Select. STEP 3 Scroll to Bluetooth Configuration and press Select. STEP 4 Scroll to Bluetooth Profiles and press the Right Arrow key.
The mobile phone appears in the list of Bluetooth devices.

The Cisco SPA IP phones can receive and display text messages via SIP according to RFC3428. Users can receive text messages. WIP310 users can send and receive text messages. When this feature is enabled, the phone displays messages up to 255 characters in length. The message appears on the phone display along with the date and time. Service providers could use text messages to: Send billing information, calling minutes consumed, minutes available Include additional text with a call to facilitate call processing
Cisco SPA303 and Cisco SPA50XG: To enable text message receipt on the Cisco SPA303 and SPA50XG phones:
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the User tab.
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STEP 4 Under Supplementary Services, in the Text Message field, choose yes. STEP 5 (Optional) To enable receipt of text messages from a third party directly without
proxy involvement, in the Text Message from 3rd Party field, choose yes.
Cisco SPA525G/525G2 To enable text messaging on the Cisco SPA525G phones:

STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the User tab. STEP 4 Under Supplementary Services, in the Display Text Message on Recv field,
choose yes.
STEP 5 (Optional) To enable receipt of text messages from a third party directly without
proxy involvement, in the Text Message from 3rd Party field, choose yes.
WIP310
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the Phone tab. STEP 4 Under SMS Enable, choose yes. STEP 5 Click Submit All Changes.
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Enabling the Web Server
Enabling the Web Server

The web server allows administrators and users to log in to the phone using a web-based configuration utility. Administrators and users have different privileges and see different options for the phone based on their role. To enable the web server:
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the System tab. STEP 4 In the Enable Web Server field, choose yes to enable the web administration
server for the phone. (For the Cisco 301 and Cisco SPA501G, this can be configured using the IVR. See the Using IVR on the Cisco SPA301 and Cisco SPA501G IP Phone section on page 29.)
STEP 5 In the Web Server Port field, enter the port to access the server. The default is port
80 for the configuration utility.

STEP 6 In the Enable Web Admin Access field, you can enable or disable local access to
the Admin Login of the configuration utility. Defaults to yes (enabled). (For the Cisco SPA301 and Cisco SPA501G, can be configured using the IVR. See the Using IVR on the Cisco SPA301 and Cisco SPA501G IP Phone section on page 29.)
STEP 7 In the Admin Passwd field, enter a password if you want the system administrator
to log in to the configuration utility with a password. The password prompt will appear when an administrator clicks Admin Login. The maximum password length is 32 characters.
STEP 8 In the User Password field, enter a password if you want users to log in to the
configuration utility with a password. The password prompt will appear users click User Login. The maximum password length is 32 characters
You can also enable the configuration utility from the Phone tab (does not apply to the WIP310):
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Configuring Lightweight Directory Access Protocol (LDAP) for the Cisco SPA300 Series and Cisco SPA500 Series IP Phones
STEP 3 Click the Phone tab. STEP 4 Under Web Serv, choose yes. STEP 5 Click Submit All Changes.
The Cisco SPA300 Series and Cisco SPA500 Series IP Phones support Lightweight Directory Access Protocol v3 to enable the retrieval of directory information. The LDAP Corporate Directory Search feature, when configured and enabled on a Cisco SPA300 Series or Cisco SPA500 Series IP Phone, allows a user to search a specified LDAP directory for a name, phone number, or both. (LDAP is not supported on the WIP310.) LDAP-based directories, such as Microsoft Active Directory 2003 and OpenLDAPbased databases, are supported. These instructions assume you have the following equipment and services: A functional LDAP server such as OpenLDAP or Microsoft's Active Directory Server 2003 A Cisco SPA300 Series or Cisco SPA500 Series IP Phone running at least 6.1.3a software on a functional network
Users access LDAP from the Directory menu on their IP phone. There is a limit of 20 records returned from an LDAP search. Before you use the LDAP Corporate Directory Search feature of your phone, you need to configure some basic information.
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the System tab. STEP 4 In the Optional Network Configuration section, under Primary DNS, enter the IP
address of the DNS server. (Only required if using Active Directory with authentication set to MD5.)
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STEP 5 In the Optional Network Configuration section, under Domain, enter the LDAP
domain. (Only required if using Active Directory with authentication set to MD5.)
NOTE Some sites may not deploy DNS internally and instead use Active Directory 2003.
In this case, it is not necessary to enter a Primary DNS address and an LDAP Domain. However, with Active Directory 2003, the authentication method is restricted to Simple.
STEP 6 Click the Phone tab. STEP 7 Under LDAP Corporate Directory Search, in the LDAP Dir Enable field, choose yes
to enable LDAP and cause the name defined in LDAP Corp Dir Name to appear in the phones Directory menu.
STEP 8 Configure values for the fields in the following table and click Submit All Changes.
Parameter LDAP Corp Dir Name
Description Enter a free-form text name, such as Corporate Directory. Enter a fully qualified domain name or IP address of LDAP server, in the following format:
nnn.nnn.nnn.nnn
LDAP Server
Enter the host name of the LDAP server if the MD5 authentication method is used. LDAP Auth Method Select the authentication method that the LDAP server requires. Choices are:
NoneNo authentication is used between the client and the server. SimpleThe client sends its fully-qualified domain name and password to the LDAP server. May present security issues. Digest-MD5The LDAP server sends authentication options and a token to the client. The client returns an encrypted response that is decrypted and verified by the server.
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Parameter LDAP Client DN
Description Enter the distinguished name domain components [dc] ; for example:
dc=cv2bu,dc=com
If using the default Active Directory schema (Name(cn)->Users->Domain), an example of the client DN follows:
cn="David Lee",dc=users,dc=cv2bu,dc=com
LDAP Username
Enter the username for a credentialed user on the LDAP server. Enter the password for the LDAP username. Specify a starting point in the directory tree from which to search. Separate domain components [dc] with a comma. For example: dc=cv2bu,dc=com
LDAP Password LDAP Search Base
LDAP Last Name Filter
This defines the search for surnames [sn], known as last name in some parts of the world. For example, sn:(sn=*$VALUE*). This search allows the provided text to appear anywhere in a name, beginning, middle, or end. You must enter a value in both the last name and first name fields so that the LDAP corporate directory option displays on the phone. If both fields are empty, the directory does not display.
LDAP First Name Filter
This defines the search for the common name [cn]. For example, cn:(cn=*$VALUE*). This search allows the provided text to appear anywhere in a name, beginning, middle, or end. You must enter a value in both the last name and first name fields so that the LDAP corporate directory option displays on the phone. If both fields are empty, the directory does not display.
LDAP Search Item 3
Additional customized search item. Can be blank if not needed. Customized filter for the searched item. Can be blank if not needed.
LDAP Item 3 Filter
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Parameter LDAP Search Item 4
Description Additional customized search item. Can be blank if not needed. Customized filter for the searched item. Can be blank if not needed. Format of LDAP results display on phone where:
LDAP Item 4 Filter
LDAP Display Attrs
aAttribute name cnCommon name snSurname (last name) telephoneNumberPhone number nDisplay name
For example, n=Phone will cause "Phone:" to be displayed in front of the phone number of an LDAP query result when the detail soft button is pressed. ttype When t=p, that is, t is of type phone number, then the retrieved number can be dialed. Only one number can be made dialable. If two numbers are defined as dialable, only the first number is used. For example, a=ipPhone, t=p; a=mobile, t=p; This example results in only the IP Phone number being dialable and the mobile number will be ignored.
pphone number
When p is assigned to a type attribute, example t=p, then the retrieved number will be dialable by the phone.
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Parameter LDAP Number Mapping
Description Can be blank if not needed. NOTE With the LDAP number mapping you can manipulate the number that was retrieved from the LDAP server. For example, you can append 9 to the number if your dial plan requires a user to enter 9 before dialing. Add the 9 prefix by adding (<:9xx.>) to the LDAP Number Mapping field. For example, 555 1212 would become 9555 1212. If you do not manipulate the number in this fashion, a user can use the Edit Dial feature to edit the number before dialing out.
The following graphic is an example of an LDAP configuration:
For more information on LDAP, including troubleshooting information, see the Configuring LDAP Directory Search on SPA SIP IP Phones Application Note, available from http://www.cisco.com/web/partners/sell/smb/products/ voice_and_conferencing.html#~vc_technical_resources (partner log on required).
Configuring BroadSoft Directory
The BroadSoft directory service enables users to search and view their personal, group, or enterprise contacts. This application feature uses BroadSoft's Extended Services Interface (XSI).
Cisco Small Business SPA300 Series, SPA500 Series, and WIP310 IP Phone Administration Guide 89

To configure the BroadSoft Directory service:

STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the Phone tab. STEP 4 Under Broadsoft Settings, configure the following:
Directory Enable: Set to yes. XSI Host Server: Enter the name of the server; for example, xsp.xdp.broadsoft.com. Directory Name: Name of the directory. Displays on the users phone as a directory choice (for example, Johns Personal Directory). Directory Type: Select the type of BroadSoft directory: Enterprise (default): Allows users to search on last name, first name, user or group ID, phone number, extension, department, or email address. Group: Allows users to search on last name, first name, user ID, phone number, extension, department, or email address. Personal: Allows users to search on last name, first name, or telephone number.
Directory UserID: BroadSoft User ID of the phone user; for example, [email protected]. Directory Password: Alphanumeric password associated with the User ID.
Configuring Synchronization of Do Not Disturb and Call Forward

Enabling synchronization of Do Not Disturb and Call Forward allows the phone to synchronize with the call server (for example, the BroadSoft server) so that if Do Not Disturb or Call Forwarding settings are changed on the phone, changes are also made on the server; if changes are made on the server, they are propagated to the phone. This feature is disabled by default.
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Configuring XML Services
To enable synchronization:
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the Phone tab. STEP 4 Under Broadsoft Settings, in the Call Feature Sync Ext field, choose the extension
(1 through 5) that is registered to the BroadSoft server.


The Cisco SPA300 Series and Cisco SPA500 Series IP Phones provide support for XML services, such as an XML Directory Service or other XML applications. The following table shows some Cisco XML objects that are supported by the Cisco SPA300 Series and Cisco SPA500 Series IP Phones:
Cisco XML Object

CiscoIPPhoneMenu CiscoIPPhoneText CiscoIPPhoneInput CiscoIPPhoneDirectory CiscoIPPhoneIconMenu CiscoIPPhoneStatus CiscoIPPhoneExecute
Supported Phone
Cisco SPA525G, Cisco SPA50X, Cisco SPA30X
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3
Supported Phone
Cisco SPA525G
Cisco XML Object

CiscoIPPhoneImage CiscoIPPhoneImageFile CiscoIPPhoneGraphicMenu CiscoIPPhoneFileMenu CiscoIPPhoneStatusFile CiscoIPPhoneResponse CiscoIPPhoneErrror CiscoIPPhoneGraphicFileMenu Init:CallHistory Key:Headset EditDial:n
Cisco SPA50X
You can use macro variables in XML URLs. The following macro variables are supported: User IDUID1, UID2 Display nameDISPLAYNAME1, DISPLAYNAME2 Auth IDAUTHID1, AUTHID2 ProxyPROXY1, PROXY2 MAC AddressMA Product NamePN Product Series NumberPSN Serial NumberSERIAL_NUMBER
For more information on XML support, see the Cisco Small Business Support community. The URL is given in Appendix C, Where to Go From Here.
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Configuring Music On Hold
To configure the phone to connect to an XML Directory service:

STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the Phone tab. STEP 4 Enter the following information:
XML Directory Service Name: Name of the XML Directory. Displays on the users phone as a directory choice. XML Directory Service URL: URL where the XML Directory is located.
To configure the phone to connect to an XML application:

STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the Phone tab. STEP 4 Enter the following information:
XML Application Service Name: Name of the XML application. Displays on the users phone as a menu item. XML Application Service URL: URL where the XML application is located.
NOTE If you have configured an unused line button to connect to an XML
application, the button connects to the URL configured here, unless you enter a different URL when configuring the line button. See the Configuring Unused Line Keys to Access Services section on page 40.
Configuring Music On Hold

Your phone can play music on hold if it is part of a system that has a music-on-hold (MOH) server. To configure music on hold:
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Configuring Extension Mobility
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the Ext <number> tab. STEP 4 Under Call Feature Settings, in the MOH Server field, enter the user ID or the URL
of the MOH streaming audio server. If you enter a user ID (no server), the current or outbound proxy is contacted. Defaults to blank (no MOH). If used with a Cisco SPA 9000 Voice System, defaults to imusic. For more information, see the Cisco SPA 9000 Voice System Administration Guide.
Configuring Extension Mobility

NOTE This feature is not available on the WIP310.
Extension mobility allows mobile users to access their personalized phone settings, such as the personal extensions, shared lines, and speed dials, from other phones. For example, people who work different shifts or who work at different desks during the week can share an extension, yet have their own personalized settings. EM is supported for BroadSoft and other servers. EM dynamically configures a phone according to the current user. A Login prompt appears on the phone display when EM is enabled on a phone (for example, a conference room phone). A user can either enter their User ID and Password to access their personal phone settings, or ignore the login and use the phone as a guest. After logging on, users have access to personal directory numbers, services, speed dials, and other properties on the phone. When a user logs out, the phone reverts to a basic profile with limited features enabled. To configure extension mobility:
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the Phone tab. STEP 4 Under Extension Mobility, in the EM Enable field, choose yes.
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Configuring Video Surveillance on the Cisco SPA525G/525G2
STEP 5 In the EM User Domain field, enter the domain for the phone, or the authentication
server. For example, "@domain.com," which is appended to the user ID ([email protected]) for authentication to the HTTP server.
You must also configure the Extension Mobility parameters in the profile rule field in the Provisioning tab. See the Provisioning Parameters for Extension Mobility on Cisco SPA500 Series IP Phones application note at: https://www.myciscocommunity.com/docs/DOC-11277 For more information on extension mobility and BroadSoft, see http://www.broadsoft.com.

The Cisco SPA525G/525G2 provides a simple video surveillance solution for a small business office. The Cisco SPA525G/525G2 works with the Cisco WVC2300 Wireless-G Business Internet Video Camera and the Cisco PVC2300 Business Internet Video Camera to provide simple video monitoring from your IP phone of a location such as a lobby entrance or doorway. Up to four cameras can be monitored from one IP phone.
NOTE Camera audio is not supported.
The Cisco SPA525G/525G2 connects to the videocamera and provides a realtime video stream display from the camera. Storage and manipulation of video and physical camera control are not available from the IP phone. The IP phone supports the camera display at a rate of two to three frames per second with good video quality. However, video quality can degrade if the camera is processing multiple streaming sessions, there is heavy Wi-Fi network traffic, or the IP phone is performing other processing. To avoid degrading voice audio quality on a call, the frame rate decreases to one frame per second if a codec other than G.711 is used for a call or when the user accesses the video monitoring page during a call. To configure the video surveillance feature, perform the steps outlined in the following sections.
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Configuring the User Name and Account on the Camera

STEP 1 Download and install the software release for the camera that provides video
monitoring support. For more information, consult the release notes for the camera software.
STEP 2 After installing the camera software, use the configuration utility to create a user ID
and password that will be used by the phone to connect to the camera. The IP phone user account that you create should have viewer privileges.
Entering Camera Information Into the Cisco SPA525G/525G2 Configuration Utility

STEP 1 Log in to the configuration utility. STEP 2 Click the User tab. STEP 3 Under Camera Settings, in the Enable Video VLAN field, choose yes. This option
separates camera traffic to a separate VLAN.

STEP 4 (Optional) If configuring Virtual LAN (VLAN) support, in the Enable Video VLAN
field, choose yes. The default Video VLAN ID is 1, the data VLAN. To separate traffic onto another VLAN (for example, a VLAN for video traffic only), enter the ID for that VLAN. (Video VLAN parameters do not apply to Wi-Fi or VPN.)
STEP 5 Under Camera Profile 1, enter the settings for the first camera. Enter the camera
name (for example, Lobby). This name is displayed on the phone display screen to identify the camera.
STEP 6 In the Access URL field, enter the URL to access the camera, in the following
format:
rtsp://xxx.xxx.x.xxx/img/jpgvideo.sav
where xxx.xxx.x.xxx is the IP address of the camera.

STEP 7 In the Access User Name field, enter the username for the phone that you created
using the cameras configuration utility.

STEP 8 In the Access Password field, enter the password for the phone username that you
created using the cameras configuration utility.

STEP 9 (Optional) In the Associated Caller ID field, enter the phone number of the phone
associated with the camera. For example, if the camera is located in the lobby, you
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may want to enter the extension of the lobby phone if one is installed there. People monitoring that camera from their phone can press Call to dial the number of the phone associated with the camera. For example, someone monitoring the lobby could call the receptionist to identify a visitor.
STEP 10 Repeat Step 4 through Step 8 for each camera. STEP 11 Click Submit All Changes.
Viewing the Video

To view video from the phone:
STEP 1 Press the Setup button. STEP 2 Scroll to Video Monitoring and press Select. STEP 3 Scroll to the camera from which you want to view and press Monitor or Select.
Pressing Call dials the number associated with the camera (see Entering Camera Information Into the Cisco SPA525G/525G2 Configuration Utility, page 96).
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4
Configuring SIP, SPCP, and NAT
The Cisco SPA IP Phones use the following protocols: Session Initiation Protocol (SIP)Cisco SPA300 Series, Cisco SPA500 Series, WIP310 Cisco Smart Phone Control Protocol (SPCP)Cisco SPA300 Series, Cisco SPA500 Series
This chapter describes how to configure the phone protocols and other parameters. It contains the following sections: Session Initiation Protocol and Cisco IP Phones, page 98 Configuring SIP, page 101 Configuring SPCP on the Cisco SPA525G/525G2, page 122 Configuring SPCP on the Cisco SPA300 Series and Cisco SPA50XG, page 123 Network Address Translation (NAT) and Cisco IP Phones, page 123

Cisco IP phones use Session Initiation Protocol (SIP), allowing interoperation with all ITSPs supporting SIP. SIP handles signaling and session management within a packet telephony network. Signaling allows call information to be carried across network boundaries. Session management controls the attributes of an end-to-end call. The following diagram shows a SIP request for connection to another subscriber in the network.
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In typical commercial IP telephony deployments, all calls go through a SIP proxy server. The requesting phone is called the SIP user agent server (UAS), while the receiving phone is called the user agent client (UAC).
SIP UA
2 4
SIP Proxy
RTP SIP Proxy

3
SIP Proxy
1
SIP UA
SIP message routing is dynamic. If a SIP proxy receives a request from a UAS for a connection but cannot locate the UAC, the proxy forwards the message to another SIP proxy in the network. When the UAC is located, the response is routed back to the UAS, and a direct peer-to-peer session is established between the two UAs. Voice traffic is transmitted between UAs over dynamically-assigned ports using Real-time Protocol (RTP). The Internet protocol RTP transmits real-time data such as audio and video; it does not guarantee real-time delivery of data. RTP provides mechanisms for the sending and receiving applications to support streaming data. Typically, RTP runs on top of the UDP protocol. See Configuring NAT Mapping with STUN section on page 125.
SIP Over TCP

To guarantee state-oriented communications, Cisco IP phones can use TCP as the transport protocol for SIP. This protocol is 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.
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TCP overcomes the problem with UDP ports being blocked by corporate firewalls. 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.
SIP Proxy Redundancy

An average SIP proxy server may handle tens of thousands of subscribers. A backup server allows an active server to be temporarily switched out for maintenance. Cisco phones support the use of backup SIP proxy servers to minimize or eliminate service disruption. A static list of proxy servers is not always adequate. If your user agents are served by different domains, for example, you would not want to configure a static list of proxy servers for each domain into every Cisco IP phone. A simple way to support proxy redundancy is to configure a SIP proxy server in the Cisco IP phone configuration profile. The DNS SRV records instruct the phones to contact a SIP proxy server in a domain named in SIP messages. The phone consults the DNS server. If configured, 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 Cisco IP phone tries to contact the hosts in the order of their priority. If the Cisco IP phone currently uses a lower-priority proxy server, the phone periodically probes the higher-priority proxy and switches to the higher-priority proxy when available.
Configuring Survivable Remote Site Telephony (SRST) Support

The proxy and outbound proxy fields in the Ext tab can be configured with an extension that includes a statically-configured DNS SRV record or DNS A record. This allows for failover and fallback functionality with a secondary proxy server. The format for the parameter value is as follows:
FQDN format: hostname[:port][:SRV=host-list OR :A=ip-list] host-list: srv[|srv[|srv]] srv: hostname[:port][:p=priority][:weight][:A=ip-list] ip-list: ip-addr[,ip-addr[,ip-addr]]
The default priority is 0 and default weight is 1. The default port is 0, and the application substitutes the proper port value (for example, port 5060 for SIP).
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Configuring SIP
RFC3311 Support
The Cisco SPA525G supports RFC3311, the SIP UPDATE Method.
Support for SIP NOTIFY XML-Service

The Cisco SPA300 Series and Cisco SPA500 Series IP Phones support the SIP NOTIFY XML-Service event. On receipt of a SIP NOTIFY message with an XMLService event, the IP phone challenges the NOTIFY with a 401 response if the message does not contain correct credentials. The client must be furnish the correct credentials using MD5 digest with the SIP account password for the corresponding line of the IP phone. The body of the message may contain the XML event Message. For example:
<CiscoIPPhoneExecute> <ExecuteItem Priority="0" URL="http://xmlserver.com/event.xml"/> </CiscoIPPhoneExecute>
Authentication:
challenge = MD5( MD5(A1) ":" nonce ":" nc-value ":" cnonce ":" qop-value ":" MD5(A2) ) where A1 = username ":" realm ":" passwd and A2 = Method ":" digest-uri
Configuring SIP
SIP settings for the Cisco SPA IP Phones are configured for the phone in general and for individual extensions. The following sections describe SIP configuration.
Configuring SIP Parameters

To configure general SIP parameters, including enabling CTI:
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the SIP tab.
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Configuring SIP
STEP 4 Under SIP Parameters, make the necessary configuration changes to the fields
shown in the following table and click Submit All Changes.
Parameter
SIP Reg User Agent Name
Description
User-Agent name used in a REGISTER request. If not specified, the SIP User Agent Name is also used for the REGISTER request. Defaults to blank. Accept-Language header used. If empty, the header is not included. Defaults to blank.
SIP Accept Language
DTMF Relay MIME Type
MIME Type used in a SIP INFO message to signal a DTMF event. This parameter must match that of the service provider. Defaults to application/dtmf-relay.
Remove Last Reg
If set to yes, removes the last registration before re-registering (if the value is different). Defaults to no. If set to yes, the Cisco IP phone uses compact SIP headers in outbound SIP messages. If inbound SIP requests contain normal (non-compact) headers, the phone substitutes incoming headers with compact headers. If set to no, the Cisco IP phone uses normal SIP headers. If inbound SIP requests contain compact headers, the phone reuses the same compact headers when generating the response, regardless of this setting. Defaults to no.
Use Compact Header
Escape Display Name
Setting this parameter to yes encloses the configured Display Name string 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. Defaults to yes. If set to yes, enables SIP for Business (supports Sylantro call flows) call features. See www.broadsoft.com for more information. Defaults to no. If set to yes enables support for the BroadSoft Talk Package, which lets users answer or resume a call by clicking a button in an external application. Defaults to no.
SIP-B Enable
Talk Package
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Configuring SIP
4
Description
If set to yes, enables support for the BroadSoft Hold Package, which lets users place a call on hold by clicking a button in an external application. Defaults to no.
Parameter
Hold Package
Conference Package
If set to yes, enables support for the BroadSoft Conference Package, which enables users to start a conference call by clicking a button in an external application. Defaults to no. If set to yes, the Cisco IP phone sends out a NOTIFY with event=conference when starting a conference call (with the BroadSoft Conference Package). Defaults to no. If set to yes, the Cisco IP phone includes SDP syntax c=0.0.0.0 when sending a SIP re-INVITE to a peer to hold the call. If set to no, the Cisco IP phone does not include the c=0.0.0.0 syntax in the SDP. With either setting, the phone includes a=sendonly syntax in the SDP. Defaults to yes.
Notify Conference
RFC 2543 Call Hold
Random REG CID On Reboot
If set to yes, the IP phone uses a different random call-ID for registration after the next software reboot. If set to no, the IP phone tries to use the same call-ID for registration after the next software reboot. With either setting the phone uses a new random call-ID for registration after a power-cycle. Defaults to no. NOTE Not applicable to the WIP310.
Mark All AVT packets
If set to yes, all audio video transport (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. Defaults to yes.
SIP TCP Port Min
Specifies the lowest TCP port number that can be used for SIP sessions. Defaults to 5060. Specifies the highest TCP port number that can be used for SIP sessions. Defaults to 5080.
SIP TCP Port Max
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Description
If set to yes, enables the computer telephony integration (CTI), where a computer can act as a call center handling all sorts of incoming and outgoing communications., including phone calls, faxes, and text messages. The CTI interface allows a third-party application to control and monitor the state of a Cisco IP phone and, for example, initiate or answer a call by clicking a mouse on a PC, NOTE CTI must be enabled on the Cisco SPA300 Series or Cisco SPA500 Series IP Phones for an attached Cisco SPA500S to properly monitor the IP phone's line status. If setting up a Cisco SPA500S, see Chapter 9, Configuring the Cisco SPA500S Attendant Console.
Parameter
CTI Enable
Defaults to no.
Caller ID Header Select where the IP phone gets its caller ID from: PAID-RPID-FROM P-ASSERTED-IDENTITY REMOTE-PARTY-ID FROM header Defaults to PAID-RPID-FROM. NOTE SRTP Method Not applicable to the WIP310.
Selects the method to use for SRTP. Two choices are available:
x-sipuralegacy SRPT method s-descriptornew method compliant with RFC-3711 and RFC-4568
The default value is "x-sipura. NOTE Hold Target Before REFER Not applicable to WIP310.
Controls whether to hold call leg with transfer target before sending REFER to the transferee when initiating a fully-attended call transfer (where the transfer target has answered). Default value is no, where the call leg is not held. NOTE Not applicable to WIP310.
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Configuring SIP Timer Values

To configure SIP timer values:
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the SIP tab. STEP 4 Under SIP Timer Values, make the necessary configuration changes to the fields
shown in the following table and click Submit All Changes.
Parameter SIP T1
Description RFC 3261 T1 value (RTT estimate). Ranges from 0 to 64 seconds. Defaults to .5 seconds.
SIP T2
RFC 3261 T2 value, which is the maximum retransmit interval for non-INVITE requests and INVITE responses. Ranges from 0 to 64 seconds. Defaults to 4 seconds.
SIP T4
RFC 3261 T4 value, which is the maximum duration a message remains in the network. Ranges from 0 to 64 seconds. Defaults to 5 seconds.
SIP Timer B
RFC 3261 INVITE transaction time-out value. Ranges from 0 to 64 seconds. Defaults to 16 seconds.
SIP Timer F
RFC 3261 Non-INVITE transaction time-out value. Ranges from 0 to 64 seconds. Defaults to 16 seconds.
SIP Timer H
RFC 3261 INVITE final response time-out value for ACK receipt. Ranges from 0 to 64 seconds. Defaults to 16 seconds.
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Description FRC 3261 wait time for response retransmits. Ranges from 0 to 64 seconds. Defaults to 16 seconds.
Parameter SIP Timer D
SIP Timer J
RFC 3261 Wait time for Non-INVITE request retransmits. Ranges from 0 to 64 seconds. Defaults to 16 seconds.
INVITE Expires
INVITE request Expires header value. If you enter 0, the Expires header is not included in the request. Ranges from 0 to 19999999999999999999999999999999. Defaults to 240 seconds.
ReINVITE Expires
ReINVITE request Expires header value. If you enter 0, the Expires header is not included in the request. Ranges from 0 to 19999999999999999999999999999999. Defaults to 30
Reg Min Expires
Minimum registration expiration time allowed from the proxy in the Expires header or as a Contact header parameter. If the proxy returns a value less than this setting, the minimum value is used. Defaults to 1 second.
Reg Max Expires
Maximum registration expiration time allowed from the proxy in the Min-Expires header. If the value is greater than this setting, the maximum value is used. Defaults to 7200 seconds.
Reg Retry Intvl (see Note below) Reg Retry Long Intvl (see Note below)
Interval to wait before the Cisco IP phone retries registration after failing during the previous registration. Do not enter 0. Defaults to 30 seconds. When registration fails with a SIP response code that does not match the Retry Reg response status code (RSC) value (see next table), the Cisco IP phone waits for this length of time before retrying. If this interval is 0, the Cisco IP phone stops trying. This value should be much larger than the Reg Retry Intvl value. Defaults to 1200 seconds.
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Description Random delay added to the Register Retry Intvl value when retrying REGISTER after a failure. Minimum and maximum random delay to be added to the short timer. Defaults to 0, which disables this feature.
Parameter Reg Retry Random Delay
Reg Retry Long Random Delay
Random delay add ed to Register Retry Long Intvl value when retrying REGISTER after a failure. Minimum and maximum random delay to be added to the long timer. Random delay range (in seconds) to add to the Register Retry Long Intvl when retrying REGISTER after a failure. Defaults to 0, which disables this feature. NOTE Not applicable to WIP310.
Reg Retry Intvl Cap
Reg_Retry_Intvl_CapMaximum value of the exponential delay. The maximum value to cap the exponential backoff retry delay (which starts at the Register Retry Intvl and doubles every retry). Defaults to 0, which disables the exponential backoff feature (that is, the error retry interval is always at the Register Retry Intvl). If this feature is enabled, the Reg Retry Random Delay is added on top of the exponential backoff delay value. NOTE Not applicable to WIP310.
Sub Min Expires
The lower limit of the REGISTER (subscribe) expires value returned from the proxy server. Defaults to 10 seconds.
Sub Max Expires
The upper limit of the REGISTER (subscribe) min-expires value returned from the proxy server in the Min-Expires header. Defaults to 7200 seconds.
Sub Retry Intvl
The retry interval when the last Subscribe request fails. Defaults to 10 seconds.
NOTE Cisco IP phones can use a RETRY-AFTER value when received from a SIP proxy
server that is too busy to process a request (503 Service Unavailable message). If the response message includes a RETRY-AFTER header, the phone waits for the specified length of time before retrying to REGISTER again. If a RETRY-AFTER header is not present, the phone waits for the value specified in the Reg Retry Interval or the Reg Retry Long Interval parameter.
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Configuring Response Status Code Handling

To configure response status code handling:
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the SIP tab. STEP 4 Under Response Status Code Handling, configure the following:
SIT1 through SIT4 RSCSIP 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. Defaults to blank. Try Backup RSCSIP response code that retries a backup server for the current request. Defaults to blank. Retry Reg RSCInterval the SPA9000 waits before re-trying registration after a failed registration. Defaults to blank.
Configuring RTP Parameters

To configure individual RTP parameters:
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the SIP tab. STEP 4 Under RTP Parameters, configure the following fields:
RTP Port MinMinimum port number for RTP transmission and reception. <RTP Port Min> and <RTP Port Max> should define a range that contains at least 10 even number ports (twice the number of lines); for example, 100 106. Defaults to 16384.
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RTP Port MaxMaximum port number for RTP transmission and reception. <RTP Port Min> and <RTP Port Max> should define a range that contains at least 10 even number ports (twice the number of lines); for example, 100 106. Defaults to 16482. RTP Packet SizePacket size in seconds, which can range from 0.01 to 0.16. Valid values must be a multiple of 0.01 seconds. Defaults to 0.030. Max RTP ICMP ErrNumber of successive ICMP errors allowed when transmitting RTP packets to the peer before the Cisco IP phone terminates the call. If the value is set to 0 (the default), the Cisco IP phone ignores the limit on ICMP errors, disabling the feature. RTCP Tx IntervalInterval for sending out RTCP sender reports on an active connection. During an active connection, the Cisco IP phone can be programmed to send out compound RTCP packet on the connection. Each compound RTP packet except the last one contains a sender report (SR) and a source description (SDES). The last RTCP packet contains an additional BYE packet. Each SR except the last one contains exactly 1 receiver report (RR); the last SR carries no RR. The SDES contains CNAME, NAME, and TOOL identifiers.: CNAME is set to User ID@Proxy NAME is set to Display Name (or Anonymous if user blocks caller ID) TOOL is set to the Vendor/Hardware-platform-software-version (such as Cisco/SPA9000-5.2.2(SCb)). The NTP timestamp used in the SR is a snapshot of the Cisco IP phones local time, not the time reported by an NTP server. If the Cisco IP phone receives a RR from the peer, it tries to compute the round trip delay and show it as the Call Round Trip Delay value in the Info section of the web GUI administration page. It can range from 0 to 255 seconds. Defaults to 0 (recommended).
No UDP ChecksumSelect yes if you want the Cisco IP phone to calculate the UDP header checksum for SIP messages. Since this involves computation load, you should keep the default value (no) to disable it. Symmetric RTPEnable symmetric RTP operation. If enabled, sends RTP packets to the source address and port of the last received valid inbound RTP packet. If disabled (or before the first RTP packet arrives) sends RTP to the destination as indicated in the inbound SDP. Defaults to no.
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Stats in BYEDetermines whether the IP phone 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> Defaults to no.
Configuring SDP Payload Types

Configured dynamic payloads are used for outbound calls only when the Cisco IP phone presents an SDP offer. For inbound calls with an SDP offer, the phone follows the callers assigned dynamic payload type. Cisco IP phones use the configured codec names in outbound SDP. For incoming SDP with standard payload types of 0-95, the Cisco IP phone ignores the codec names. For dynamic payload types, the Cisco IP phone identifies the codec by the configured codec names (comparison is case-sensitive). To configure SDP payload types:
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the SIP tab. STEP 4 Under SDP Payload Types, configure the fields shown in the following table and
click Submit All Changes.
Parameter AVT Dynamic Payload
Description AVT dynamic payload type. Ranges from 96-127. Defaults to 101.
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Description This parameter defines the Codec Number used in the SIP messaging for the Dynamic Payload size mechanism. This number should match the number configured in the network/other party to enable the use of Dynamic Payload. The best range is 96-127 for any dynamic payload type. Defaults to blank. G.726-16 dynamic payload type. Ranges from 96-127. Defaults to 98. NOTE Not applicable to Cisco SPA525G.
Parameter INFOREQ Dynamic Payload
G726r16 Dynamic Payload
G.726-24 dynamic payload type. Ranges from 96-127. Defaults to 97. NOTE Not applicable to Cisco SPA525G.
G726r32 dynamic payload type. The default is 2.
G.726-40 dynamic payload type. Ranges from 96-127. Defaults to 96. NOTE Not applicable to Cisco SPA525G.
G729b Dynamic Payload
G729b Dynamic Payload type. Defaults to 99.
EncapRTP Dynamic Payload
EncapRTP Dynamic Payload type. Defaults to 112.
RTP-Start-Loopback Dynamic
RTP-Start-Loopback Dynamic Payload. Defaults to 113.
RTP-Start-Loopback Codec
RTP-Start-Loopback Codec. Select one of following: G711u, G711a, G726-16, G726-24, G726-32, G726-40, G729a, or G723. Defaults to G711u.
AVT Codec Name
AVT codec name used in SDP. Defaults to telephone-event.
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Description G.711u codec name used in SDP. Defaults to PCMU.
Parameter G711u Codec Name
G711a Codec Name
G.711a codec name used in SDP. Defaults to PCMA.
G726r16 Codec Name
G.726-16 codec name used in SDP. Defaults to G726-16. NOTE Not applicable to Cisco SPA525G.
G726r24 Codec Name
G726r32 Codec Name
G.726-32 codec name used in SDP. Defaults to G726-32.
G726r40 Codec Name
G729a Codec Name
G.729a codec name used in SDP. Defaults to G729a.
G729b Codec Name
G.729b codec name used in SDP. Defaults to G729ab.
G723 Codec Name
G.723 codec name used in SDP. Defaults to G723. NOTE Not applicable to the WIP310 or Cisco SPA525G.
EncapRTP Codec Name
EncapRTP codec name used in SDP. Defaults to encaprtp.
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Configuring SIP
Configuring SIP Settings for Extensions

STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the Extension <number> tab. STEP 4 Under Network Settings, configure the following fields:
Parameter SIP ToS/DiffServ Value
Description Time of service (ToS)/differentiated services (DiffServ) field value in UDP IP packets carrying a SIP message. Defaults to 0x68.
SIP CoS Value [0-7]
Class of service (CoS) value for SIP messages. Defaults to 3.
RTP ToS/DiffServ Value RTP CoS Value [0-7]
ToS/DiffServ field value in UDP IP packets carrying RTP data. Defaults to 0xb8. CoS value for RTP data. Defaults to 6.
Network Jitter Level
Determines how jitter buffer size is adjusted by the SPA9000. 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. Defaults to high.
Jitter Buffer Adjustment
Controls how the jitter buffer should be adjusted. Select the appropriate setting: up and down, up only, down only, or disable. Defaults to up and down.
STEP 5 Under SIP Settings, configure the following fields:
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Configuring SIP
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Description Select from UDP, TCP, or TLS. Defaults to UDP.
Parameter SIP Transport
SIP Port
Port number of the SIP message listening and transmission port. Defaults to 5060.
SIP 100REL Enable
To enable the support of 100REL SIP extension for reliable transmission of provisional responses (18x) and use of PRACK requests, select yes. Otherwise, select no. Defaults to no.
EXT SIP Port Auth Resync-Reboot
The external SIP port number. If this feature is enabled, the Cisco IP phone authenticates the sender when it receives a NOTIFY message with the following requests:
resync reboot report restart XML-service
To use this feature, select yes. Otherwise, select no. Defaults to yes. SIP Proxy-Require The SIP proxy can support a specific extension or behavior when it sees this header from the user agent. If this field is configured and the proxy does not support it, it responds with the message, unsupported. Enter the appropriate header in the field provided. To use the Remote-Party-ID header instead of the From header, select yes. Otherwise, select no. Defaults to yes.
SIP Remote-Party-ID
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Description Controls when the Cisco IP phone sends BYE to terminate stale call legs upon completion of call transfers. Multiple delay settings (Referror, Refer Target, Referee, and Refer-To Target) are configured on this screen. For the Referror Bye Delay, enter the appropriate period of time in seconds. Defaults to 4.
Parameter Referror Bye Delay
Refer-To Target Contact Referee Bye Delay
To contact the refer-to target, select yes. Otherwise, select no. Default: no. For the Referee Bye Delay, enter the appropriate period of time in seconds. Defaults to 0.
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Configuring SIP
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Description SIP messages are received at or sent from the proxy listen port. This feature controls which SIP messages to log. Choices are as follows:
Parameter SIP Debug Option
noneNo logging. 1-lineLogs the start-line only for all messages. 1-line excl. OPTLogs the start-line only for all messages except OPTIONS requests/responses. 1-line excl. NTFYLogs the start-line only for all messages except NOTIFY requests/responses. 1-line excl. REGLogs the start-line only for all messages except REGISTER requests/responses. 1-line excl. OPT|NTFY|REGLogs the start-line only for all messages except OPTIONS, NOTIFY, and REGISTER requests/responses. fullLogs all SIP messages in full text. full excl. OPTLogs all SIP messages in full text except OPTIONS requests/responses. full excl. NTFYLogs all SIP messages in full text except NOTIFY requests/responses. full excl. REGLogs all SIP messages in full text except REGISTER requests/responses. full excl. OPT|NTFY|REGLogs all SIP messages in full text except for OPTIONS, NOTIFY, and REGISTER requests/ responses.
Defaults to none. Refer Target Bye Delay For the Refer Target Bye Delay, enter the appropriate period of time in seconds. Defaults to 0. Sticky 183 If this feature is enabled, the IP telephony ignores further 180 SIP responses after receiving the first 183 SIP response for an outbound INVITE. To enable this feature, select yes. Otherwise, select no. Defaults to no.
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Description When enabled, authorization is required for initial incoming INVITE requests from the SIP proxy. NOTE Not applicable to the WIP310.
Parameter Auth INVITE
Ntfy Refer On 1xxTo-Inv
If set to yes, as a transferee, the phone will send a NOTIFY with Event:Refer to the transferor for any 1xx response returned by the transfer target, on the transfer call leg. If set to no, the phone will only send a NOTIFY for final responses (200 and higher). NOTE Not applicable to the WIP310.
Use Anonymous with RPID
This parameter applies only if <SIP Remote-Party-ID> is set to yes; otherwise, it is ignored. If the parameter is set to yes, the FROM header's display-name and user-id fields are set to anonymous when the caller blocks his caller-id. If the parameter is set to no, the FROM header's display-name and user-id are not masked. The Remote-PartyID header indicates privacy=full when the caller wishes to block his caller-id. Default: yes. NOTE Not applicable to the WIP310.
Set G729 annexb
Configure G.729 Annex B settings. NOTE Not applicable to the Cisco SPA525G.
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Description For configuration of a SIP event package, SIP PUBLISH, that enables the collection and reporting of metrics that measure the quality for VoIP sessions. Voice call quality information derived from RTCP-XR and call information from SIP is conveyed from a User Agent in a session to the third party in SIP PUBLISH method. To configure, first configure RTCP-XR (see the Configuring RTP Parameters section on page 108). For example, you need to configure the RTCP Tx Interval. In the Voice Quality Report Address field, enter the name of the collector that will collect the statistics from the SIP PUBLISH events. For example, enter collector@fully-qualified-domain-name ([email protected]) or collector@IP-address ([email protected]). After RTCP-XR feature is enabled, the call status information is updated on the Voice > Info page during an active call. Additionally, RTCP-XR packets containing a voice metrics block report will be sent with the interval specified in the RTCP Tx Interval. When the call session is ended, a SIP PUBLISH with voice metrics info is sent to the collector endpoint.
Parameter Voice Quality Report Address
Configuring a SIP Proxy Server

To configure SIP proxy and registration parameters:
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the Extension <number> tab. STEP 4 Configure the proxy and registration parameters for each extension.
Parameter Proxy
Description SIP proxy server and port number set by the service provider for all outbound requests. For example: 192.168.2.100:6060. NOTE Port number is optional. The default is port 5060.
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Description Enables an outbound proxy (for example, 172.20.2.1:5060port is optional) or a domain name such as sip.server.com as long as this name is a fullyqualified domain name. If set to no, the Outbound Proxy and Use OB Proxy in Dialog fields are ignored. Defaults to no. Optionally, the proxy can be configured (Cisco SPA300 or Cisco SPA500 series only) for Survivable Remote Site Telephony (SRST) support. The proxy is configured with an extension that includes a statically-configured DNS SRV record or DNS A record. Configuring the proxy allows for failover and fallback functionality with a secondary proxy server. For example: For SRV Record: sip.server.com:SRV=node1.sip.server.com:5060:p=1:w=5 0|node2.sip.server.com:5060:p=2:w=50 NOTE Set "Use DNS SRV" to no and "DNS SRV Auto Prefix" to no. For A Record: sip.server.com:A=172.20.2.1,172.20.2.2 NOTE Set "Use DNS SRV" to no and "DNS SRV Auto Prefix" to no.
Parameter Use Outbound Proxy
Outbound Proxy
SIP outbound proxy server where all outbound requests are sent as the first hop. Select yes for SIP requests to be sent to the outbound proxy within a dialog. This field is ignored if:
Use OB Proxy In Dialog
or
Use Outbound Proxy is set to no
Outbound Proxy is blank
Defaults to yes. Register Enables periodic registration with the proxy. This parameter is ignored if a proxy is not specified. Defaults to yes.
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Description Enables making outbound calls without successful (dynamic) registration by the phone. If set to no, the dial tone plays only when registration is successful. Defaults to no.
Parameter Make Call Without Reg
Register Expires
Defines how often the phone renews registration with the proxy. If the proxy responds to a REGISTER with a lower expires value, the phone renews registration based on that lower value instead of the configured value. If registration fails with an Expires too brief error response, the phone retries with the value specified in the Min-Expires header of the error. Defaults to 60 seconds.
Ans Call Without Reg
If enabled, the user does not have to be registered with the proxy to answer calls. Defaults to no.
Use DNS SRV
Enables DNS SRV lookup for the proxy and outbound proxy. Defaults to no.
DNS SRV Auto Prefix
Enables the phone to automatically prepend the proxy or outbound proxy name with _sip._udp when performing a DNS SRV lookup on that name. Defaults to no.
Proxy Fallback Intvl
Sets the delay after which the phone retries from the highest priority proxy (or outbound proxy) after it has failed over to a lower priority server. The phone should have the primary and backup proxy server list via DNS SRV record lookup on the server name. It needs to know proxy priority; otherwise, it does not retry. Defaults to 3600 seconds.
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Description Select Normal or Based on SRV port. The phone creates an internal list of proxies returned in the DNS SRV records. If you select Normal, the list contains proxies ranked by weight and priority. If you select Based on SRV, the phone uses normal, then inspects the port number based on the first-listed proxy port. Defaults to Normal.
Parameter Proxy Redundancy Method
Configuring Subscriber Information Parameters

To configure subscriber information parameters for each extension:
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the Ext <number> tab. STEP 4 Enter the subscriber information:
Parameter Display Name User ID Password
Description Display name for caller ID. Extension number for this line. Password for this line. Defaults to blank.
Use Auth ID
To use the authentication ID and password for SIP authentication, select yes. Otherwise, select no to use the user ID and password. Defaults to no.
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Configuring SPCP on the Cisco SPA525G/525G2
4
Description Authentication ID for SIP authentication. Defaults to blank.
Parameter Auth ID
Mini Certificate
Base64 encoded of Mini-Certificate concatenated with the 1024-bit public key of the CA signing the MC of all subscribers in the group. Defaults to blank.
SRTP Private Key
Base64 encoded of the 512-bit private key per subscriber for establishment of a secure call. Defaults to blank.
Configuring SPCP on the Cisco SPA525G/525G2

The Cisco SPA525G/525G2 can be used as part of a Cisco Unified Communications System. This system uses SPCP (also called SCCP) for call control features. To configure SPCP on the Cisco SPA525G/525G2:
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the System tab. STEP 4 Under System Configuration, in the SPA525-protocol field, choose SCCP. STEP 5 (Optional) To configure the phone to automatically detect the protocol being used
on the network to which it is connected, in the SPA525-auto-detect-sccp field, choose yes.

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Configuring SPCP on the Cisco SPA300 Series and Cisco SPA50XG
Configuring SPCP on the Cisco SPA300 Series and Cisco SPA50XG

The Cisco SPA300 Series and Cisco SPA50XG can be used as part of a Cisco Unified Communications System. This system uses SPCP (also called SCCP) for call control features. To configure SPCP on the Cisco SPA300 Series or Cisco SPA50XG:
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the System tab. STEP 4 Under System Configuration, in the Signaling Protocol field, choose SCCP. STEP 5 (Optional) To configure the phone to automatically detect the protocol being used
on the network to which it is connected, in the SPCP Auto-detect field, choose yes.
NOTE For the Cisco SPA301 and Cisco SPA501G, can be configured using the IVR. See
the Using IVR on the Cisco SPA301 and Cisco SPA501G IP Phone section on page 29.
Network Address Translation (NAT) and Cisco IP Phones

NAT is a function that allows multiple devices to share the same public, routable, IP address to establish connections over the Internet. NAT is present in many broadband access devices to translate public and private IP addresses. To enable VoIP to co-exist with NAT, some form of NAT traversal is required.
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Some ITSPs provide NAT traversal, but some do not. If your ITSP does not provide NAT traversal, you have several options. NAT Mapping with Session Border Controller, page 124 NAT Mapping with SIP-ALG Router, page 124 Configuring NAT Mapping with a Static IP Address, page 124 Configuring NAT Mapping with STUN, page 125
NAT Mapping with Session Border Controller

It is strongly recommended that you choose an ITSP that supports NAT mapping through a Session Border Controller. With NAT mapping provided by the ITSP, you have more choices in selecting a router.
NAT Mapping with SIP-ALG Router

If the ITSP network does not provide a Session Border Controller functionality, you can achieve NAT mapping by using a router that has a SIP ALG (Application Layer Gateway). By using a SIP-ALG router, you have more choices in selecting an ITSP.
Configuring NAT Mapping with a Static IP Address

If the ITSP network does not provide a Session Border Controller functionality, and if other requirements are met, you can configure NAT mapping to ensure interoperability with the ITSP. Requirements You must have an external (public) IP address that is static. The NAT mechanism used in the router must be symmetric. See Determining Whether the Router Uses Symmetric or Asymmetric NAT, on page 126.
NOTE Use NAT mapping only if the ITSP network does not provide a Session Border
Controller functionality.
STEP 1 Click Admin Login and advanced. STEP 2 Click the SIP tab. STEP 3 Under NAT Support Parameters, configure the following:
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Handle VIA received, Insert VIA received, Substitute VIA Addr: yes Handle VIA rport, Insert VIA rport, Send Resp To Src Port: yes EXT IP: Enter the public IP address for your router.
STEP 4 Click the Ext <number> tab. Configure the following:
NAT Mapping Enable: Choose yes. NAT Keep Alive Enable: Choose yes (optional).
STEP 5 Click Submit All Changes. NOTE You also need to configure the firewall settings on your router to allow SIP traffic. See Configuring SIP, on page 101.
Configuring NAT Mapping with STUN

If the ITSP network does not provide a Session Border Controller functionality, and if other requirements are met, it is possible to use STUN as a mechanism to discover the NAT mapping. This option is considered a practice of last resort and should be used only if the other methods are unavailable. Requirements STUN is a viable option only if your router uses asymmetric NAT. See Determining Whether the Router Uses Symmetric or Asymmetric NAT, on page 126. You must have a computer running STUN server software. You can use a public STUN server or set up your own STUN server.
NOTE Use NAT mapping only if the ITSP network does not provide a Session Border
Controller functionality.
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the SIP tab. STEP 4 Under NAT Support Parameters, configure the following:
Handle VIA received: yes Handle VIA rport: yes
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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.
STEP 5 Click the Ext <number> tab. Configure the following:
NAT Mapping Enable: Choose yes. NAT Keep Alive Enable: Choose yes (optional).
NOTE Your ITSP may require the phone to send NAT keep alive messages to
keep the NAT ports open permanently. Check with your ITSP to determine the requirements.
STEP 6 Click Submit All Changes. NOTE You also need to configure the firewall settings on your router to allow SIP traffic. See Configuring SIP, on page 101.
Determining Whether the Router Uses Symmetric or Asymmetric NAT

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.
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STEP 1 Make sure you do not have firewall running on your PC that could block the syslog
port (by default this is 514).

STEP 2 Log in to the configuration utility. For information about this, see the Using the
Web-Based Configuration Utility section on page 22.

STEP 3 Click Admin Login > Advanced. (For WIP310, click Admin Login.) STEP 4 Click the System tab, then set Debug Server to the IP address and port number of
your syslog server. Note that this address and port number has to be reachable from the Cisco IP phone. This port number appears on the output file name. The default port number is 514. The default output is named syslog.514.log (if port number was not specified).
STEP 5 Set Debug Level to 3. Do not change the value of the Syslog Server parameter. STEP 6 To capture SIP signaling messages, click the Ext tab. STEP 7 Set SIP Debug Option to Full. STEP 8 To collect information about what type of NAT your router uses click the SIP tab
and scroll to NAT Support Parameters.

STEP 9 Back in the SIP tab, select yes in the STUN Test Enable drop-down box.
View the debug messages to determine if your network uses symmetric NAT. Look for the Warning header in REGISTER messages, for example, Warning: 399 Spa Full Cone NAT detected.
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5
Configuring Security, Quality, and Network Features
This chapter describes how to configure security, quality, and network features for the phone. It contains the following sections: Setting Security Features, page 128 Ensuring Voice Quality, page 132 Configuring Voice Codecs, page 136 Configuring Domain and Internet Settings, page 140 Setting Optional Network Parameters, page 143 Configuring VLAN Settings for IP Phones, page 145 Configuring SSL VPN on the Cisco SPA525G/525G2, page 155

The following features help ensure that calls are secure and authenticated. SIP Initial INVITE and MWI Challenge, page 129 SIP Over TLS, page 129 SRTP and Securing Calls, page 130
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SIP Initial INVITE and MWI Challenge

SIP INVITE (initial) and MWI message in a session can be challenged by the endpoint. The purpose of this challenge is to restrict the SIP servers that are permitted to interact with the devices on the service provider network, which significantly increases the security of the VoIP network by preventing malicious attacks against the device. To configure SIP INVITE challenge:
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click Ext <number>, then scroll to the SIP Settings section. STEP 4 In the Auth INVITE field, choose yes. STEP 5 Click Submit All Changes.
SIP Over TLS

Transport layer security (TLS) is a standard protocol for securing and authenticating communications over the Internet. SIP Over TLS eliminates the possibility of malicious activity by encrypting the SIP messages by the SIP proxy of the service provider and the end user. SIP Over TLS relies on the widely-deployed and standardized Transport Layer Security (TLS) protocol. Note that SIP Over TLS encrypts only the signaling messages and not the media. A separate secure protocol such as Secure Real-Time Transport Protocol (SRTP) (see below) can be used to encrypt voice packets. The TLS protocol has two layers: TLS Record Protocol -- layered on top of a reliable transport protocol, such as SIP or TCH, it ensures that the connection is private by using symmetric data encryption and it ensures that the connection is reliable. TLS Handshake Protocol -- allows authentication between the server and client and the negotiation of an encryption algorithm and cryptographic keys before the application protocol transmits or receives any data.
TLS is application protocol-independent. Higher-level protocols such as SIP can layer on top of the TLS protocol transparently.
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The IP phones use UDP as a standard for SIP transport, but they also support SIP over TLS for added security. To enable TLS for the phone:
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click Ext <number>, then scroll to the SIP Settings section. STEP 4 Select TLS from the SIP Transport drop-down box. STEP 5 Click Submit All Changes.
SRTP and Securing Calls

Secure Real-Time Transport Protocol (SRTP) is a secure protocol for transporting real-time data over networks. Cisco SPA IP Phones use SRTP to securely send and receive real-time voice traffic from other phones and gateways. Security Description (RFC 4568) is supported. SRTP provides media encryption to ensure that media streams between devices are secure and that only the intended devices receive and read the data. When a call is secure, the voice conversation is encrypted so that others cannot eavesdrop on the conversation. To enable this feature the IP phone must have a mini-certificate installed. The supplementary service Secure All Calls (*16)Defaults to prefer to use encrypted media (voice codecs). Audio packets in both directions of outbound calls are encrypted using SRTP. To use Secure Call on an extension, you must configure Mini Certificate and SRTP Private Key for that extension. These parameters appear on the Ext tabs. Secure Call Service activates secure encryption of RTP streams between the two endpoints. You can disable this if the other endpoint (or gateway) does not support this proprietary method.
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To enable the secure call service:

STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the Phone tab. STEP 4 In the Supplementary Services section, under Secure Call Serv, choose yes. STEP 5 Click Submit All Changes.
NOTE This feature can also be configured from the User tab, under Supplementary
Services. Users can enter *18 to Secure Next CallUses encrypted media for the next outbound call (on this call appearance only). This star code is redundant if all outbound calls are secure by default. The phone can be configured for secure provisioning using the factory-installed security certificate. To determine if the Client Certificate is installed on the phone: SPA50XG: Press the Setup button and select Product Info. Scroll to Client Cert. SPA525G/525G2: Press the Setup button and select Status. Select Product Information. Scroll to Certificate. WIP310: Log in to the configuration utility. In the Info tab, under Product Information, certificate information is listed in the Client Certificate field.

This tone is 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. You can configure it on the Regional web page under Call Progress Tones. Defaults to 397@-19,507@-19;15(0/2/0,.2/.1/1,.1/2.1/2)
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Voice quality perceived by the subscribers of the IP Telephony service should be indistinguishable from that of the PSTN. Cisco IP phones support several codecs. See: Supported Codecs section on page 132 Bandwidth Requirements section on page 133 Factors Affecting Voice Quality section on page 134
Supported Codecs
Negotiation of the optimal voice codec sometimes depends on the ability of the Cisco IP Phone to match a codec name with the far-end device/gateway codec name. Cisco IP phones allow the network administrator to individually name the various codecs that are supported such that the correct codec successfully negotiates with the far-end equipment. Note that Cisco IP phones support voice codec priority. You can select up to three preferred codecs. The administrator can select the low-bit-rate codec used for each line. G.711a and G.711u are always enabled. The following table shows the codecs supported by Cisco IP phones. The third column shows the voice quality Mean Opinion Score (MOS), with a scale of 15, in which higher is better.
Codec (Voice Compression Algorithm) G.711 (A-law and u-law)
Complexity and Description
MOS Score
Very low complexity. Supports uncompressed 64 kbps digitized voice transmission 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.
4.5 Highest voice quality
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5
MOS Score
Codec (Voice Compression Algorithm) G.726
Complexity and Description
Low complexity. Supports compressed 16, 24, 32, and 40 kbps digitized voice transmission at one through ten 10 ms voice frames per packet. When no static payload value is assigned per RFC 1890, Cisco IP phones can support dynamic payloads for G.726. NOTE G.726 is supported only for 32kbps on the SPA525G.
4.1 (32 kbps)
G.729 and G.729A
G.729A low-medium complexity. G.729 medium complexity. G.729A requires about half the processing power of G.729. The G.729 and G.729A bit streams are compatible and interoperable, but not identical.
G.723.1
High complexity. Cisco IP phones support the use of ITU G.723.1 audio codec at 6.4 kbps. Up to two channels of G.723.1 can be used simultaneously. For example, Line 1 and Line 2 can be using G.723.1 simultaneously, or Line 1 or Line 2 can initiate a threeway conference with both call legs using G.723.1. NOTE G.723.1 is not supported on the 525G or WIP310.
3.8
G.722
Only one G.722 call at a time is allowed. If a conference call is placed, a SIP re-invite message is sent to switch the calls to narrowband audio. NOTE Not supported on the WIP310.
4.3 (approx)
Bandwidth Requirements
Depending on how you have your IP phones configured, each call requires 55 to 110 kbps in each direction. Therefore, using G.729 as the voice codec setting, and with an average business-grade broadband Internet connection supporting 1.5 Mbps downstream and 384 kbps upstream, a total of seven (7) simultaneous conversations can be reliably supported with adequate bandwidth available for file downloads. Cisco recommends using the Cisco IP phones with QoS-capable networking equipment that can prioritize the VoIP application traffic. QoS features are available on many data networking switches and routers. A QoS-enabled router prioritizes the packets going upstream to the ISP.
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The following table approximates the bandwidth budget for each side of the conversation (in each direction) using different codecs and number of calls. This table is based on the following assumptions: Bandwidth calculated with no silence suppression 20 millisecond of payload per RTP packet
Codec
Est. Bandwidth Budget

110 kbps 110 kbps 87 kbps 79 kbps 71 kbps 63 kbps 55 kbps
2 Calls
4 Calls
6 Calls
8 Calls
G.711 G.722 G.726-40 G.726-32 G.726-24 G.726-16 G.729
NOTE The use of silence suppression can reduce the average bandwidth budget by 30%
or more. For more information about bandwidth calculation, refer to the following websites: http://www.erlang.com/calculator/lipb/ http://www.packetizer.com/voip/diagnostics/bandcalc.html
Factors Affecting Voice Quality

The following factors contribute to voice quality: Audio compression algorithmSpeech signals are sampled, quantized, and compressed before they are packetized and transmitted to the other end. For IP Telephony, speech signals are usually sampled at 8000 samples per second with 1216 bits per sample. The compression algorithm plays a large role in determining the voice quality of the reconstructed speech signal at the other end. Cisco IP phones support the most popular audio compression algorithms for IP Telephony: G.711 a-law and u-law, G.726,
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G.729a, G.722 (not supported on WIP310) and G.723.1. (not supported on the SPA525G or WIP310.) The encoder and decoder pair in a compression algorithm is known as a codec. The compression ratio of a codec is expressed in terms of the bit rate of the compressed speech. The lower the bit rate, the smaller the bandwidth required to transmit the audio packets. Although voice quality is usually lower with a lower bit rate, it is usually higher as the complexity of the codec gets higher at the same bit rate. Silence suppressionCisco IP phones apply silence suppression so that silence packets are not sent to the other end to conserve more transmission bandwidth. IP bandwidth is used only when someone is speaking. Voice activity detection (VAD) with silence suppression is a means of increasing the number of calls supported by the network by reducing the required bidirectional bandwidth for a single call. A noise level measurement is sent periodically during silence suppressed intervals so that the other end can generate artificial comfort noise (comfort noise generator, or CNG). Packet lossAudio packets are transported by UDP, which does not guarantee the delivery of the packets. Packets may be lost or contain errors that can lead to audio sample drop-outs and distortions and lower the perceived voice quality. The Cisco SPA IP Phones apply an error concealment algorithm to alleviate the effect of packet loss. Network jitterThe IP network can induce varying delay of received packets. The RTP receiver in Cisco IP phones keeps a reserve of samples to absorb the network jitter, instead of playing out all the samples as soon as they arrive. This reserve is known as a jitter buffer. The bigger the jitter buffer, the more jitter it can absorb, but this also introduces bigger delay. Jitter buffer size should be kept to a relatively small size whenever possible. If jitter buffer size is too small, many late packets may be considered as lost and thus lowers the voice quality. Cisco IP phones dynamically adjust the size of the jitter buffer according to the network conditions that exist during a call. 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. Defaults to high.
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Configuring Voice Codecs
Jitter Buffer AdjustmentControls how the jitter buffer should be adjusted. Select the appropriate setting: up and down, up only, down only, or disable. Defaults to up and down. EchoImpedance mismatch between the telephone and the IP Telephony gateway phone port can lead to near-end echo. Cisco IP phones have a near-end echo canceller with at least 8 ms tail length to compensate for impedance match. Cisco IP phones implement an echo suppressor with comfort noise generator (CNG) so that any residual echo is not noticeable. Hardware noiseCertain levels of noise can be coupled into the conversational audio signals because of the hardware design. The source can be ambient noise or 60 Hz noise from the power adaptor. The Cisco hardware design minimizes noise coupling. End-to-end delayEnd-to-end delay does not affect voice quality directly but is an important factor in determining whether IP phone subscribers can interact normally in a conversation. A reasonable delay should be about 50 100 ms. End-to-end delay larger than 300 ms is unacceptable to most callers. Cisco IP phones support end-to-end delays well within acceptable thresholds. Adjustable Audio Frames Per PacketAllows you to set the number of audio frames contained in one RTP packet. Packets can be adjusted to contain from 110 audio frames. Increasing the number of frames decreases the bandwidth utilized, but it also increases delay and can affect voice quality.

A codec resource is considered allocated if it has been included in the SDP codec list of an active call, even though it eventually might not be chosen for the connection. If the G.729a codec is enabled and included in the codec list, that resource is tied up until the end of the call whether or not the call actually uses G.729a. If the G729a resource is already allocated (and since only one G.729a resource is allowed per IP phone), no other low-bit-rate codec can be allocated for subsequent calls. The only choices are G711a and G711u. Since two G.723.1/G.726 resources are available per IP phone, you should disable the use of G.729a to guarantee support for two simultaneous G.723/G.726 codecs.
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To configure the voice codecs on each extension:

STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the Ext <number> tab for the extension you want to configure. STEP 4 Under Audio Configuration, configure the following parameters:
Parameter Preferred Codec
Description Preferred codec for all calls. (The actual codec used in a call still depends on the outcome of the codec negotiation protocol.) Select one of the following:
G711u (all models) G711a (all models) G726-16 (not supported on WIP310, SPA525G/525G2) G726-24 (not supported on WIP310, SPA525G/525G2) G726-32 G726-40 (not supported on WIP310, SPA525G/525G2) G729a G723 (not supported on WIP310, SPA525G/525G2) G722 (not supported on WIP310)
See Supported Codecs section on page 132.

Defaults to G711u. Use Pref Codec Only To use only the preferred codecs for all calls, select yes. (The call fails if the far end does not support these codecs.) Otherwise, select no. Defaults to no. Second Preferred Codec If the first codec fails, this codec is tried. Defaults to unspecified. NOTE Not applicable to the WIP310.
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Parameter Third Preferred Codec
Description If the second codec fails, this codec is tried. Defaults to unspecified. NOTE Not applicable to the WIP310.
G729a Enable
To enable the use of the G.729a codec at 8 kbps, select yes. Otherwise, select no. Defaults to yes.
G722 Enable
Enables use of the G.722 codec. Defaults to yes. NOTE Not applicable to the WIP310.
G723 Enable
To enable the use of the G.723a codec at 6.3 kbps, select yes. Otherwise, select no. Defaults to yes. NOTE Not applicable to the WIP310, SPA525G/525G2, or Cisco SPA300 Series.
G726-16 Enable
To enable the use of the G.726 codec at 16 kbps, select yes. Otherwise, select no. Defaults to yes. NOTE Not applicable to the WIP310 or SPA525G/525G2.
G726-24 Enable
G726-32 Enable
To enable the use of the G.726 codec at 32 kbps, select yes. Otherwise, select no. Defaults to yes.
G726-40 Enable
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Parameter Release Unused Codec
Description Allows the release of codecs not used after codec negotiation on the first call so that other codecs can be used for the second line. To use this feature, select yes. Defaults to yes.
DTMF Process AVT
Select yes to process RTP DTMF events. Otherwise, select no. If this parameter is set to no, the AVT payload type is not included in outbound SDP. Defaults to yes.
Silence Supp Enable
To enable silence suppression so that silent audio frames are not transmitted, select yes. Otherwise, select no. See Ensuring Voice Quality section on page 132. Defaults to no.
DTMF Tx Method
Select the method to transmit DTMF signals to the far end: InBand, audio video transport (AVT), INFO, Auto, InBand+INFO, or AVT+INFO. InBand sends DTMF using the audio path. AVT sends DTMF as AVT events. INFO uses the SIP INFO method. Auto uses InBand or AVT based on the outcome of codec negotiation. Defaults to Auto.
DTMF Tx Volume for AVT Packet
Allows you to manually configure the AVT Tx volume. The value of this parameter is inserted into the volume field of the payload in the AVT packet. Values are based on the AVT specification as described in RFC 2833, RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals. According to RFC 2833, the volume field is represented by 6 bits, and describes the power level of the tone, expressed in dBm0 after dropping the sign. Valid range for this parameter is 0 to 63. If the provisioned value is negative, it will be negated first. Thereafter, if the value is beyond the high limit of 63, it will be clipped to 63. The default value is 0, and is the recommended setting. However, some gateways do not accept this volume setting. If the gateway does not accept the value of 0, the DTMF tone is not relayed to the remote end. As a workaround for the phone to interoperate with those gateways, you can change the value to a value greater than 0.
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Configuring Domain and Internet Settings
Parameter
Description If set to yes, the phone communicates using the remote phones preferred codec. If set to no, the Cisco IP phone communicates using its own preferred codec (as indicated in the Preferred Codec field and in the SDP by order of preferences). The default vale is no. When set to Default, the Cisco IP phone responds to an Invite with a 200 OK response advertising the preferred codec only. When set to List All, the Cisco IP phone responds listing all the codecs that the phone supports. The default value is Default, or to respond with the preferred codec only.
Use Remote Pref Codec
Codec Negotiation

Configuring Restricted Access Domains
You can configure restricted access domains. If you enter domains, the Cisco IP phones respond to SIP messages only from the entered servers.
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the System tab. STEP 4 Enter fully-qualified domain names (FQDNs) for each SIP server you want the
phone to respond to. Separate with semicolons.

Configuring DHCP, Static IP, and PPPoE Information

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STEP 3 Click the System tab. STEP 4 Configure the Internet Connection Type. Choose one of the following:
Dynamic Host Configuration Protocol (DHCP): Configure the phone to receive an IP address from the network DHCP. Cisco IP phones typically operate in a network where a DHCP server assigns the device its IP address. Because IP addresses are a limited resource, the DHCP server periodically renews the device lease on the IP address. If a phone loses its IP address for any reason, or if some other device on the network is assigned its IP address, the communication between the SIP proxy and the phone is either severed or degraded. Whenever an expected SIP response is not received within a programmable amount of time after the corresponding SIP command is sent, the DHCP Timeout on Renewal feature causes the device to request a renewal of its IP address. If the DHCP server returns the IP address that it originally assigned to the phone, the DHCP assignment is presumed to be operating correctly. Otherwise, the phone resets to try to fix the issue. Static IPConfigure a static IP address for the phone. PPPoEPoint-to-Point Protocol over Ethernet (PPPoE) relies on two widely accepted standards: PPP and Ethernet. PPPoE is a specification for connecting users on an Ethernet to the Internet through a common broadband medium, such as a single DSL line, wireless device, or cable modem. All users on an Ethernet share a common connection, so the Ethernet principles supporting multiple users in a LAN combine with the principles of PPP, which apply to serial connections.
NOTE PPPoE is only applicable to the Cisco SPA525G/525G2.
Setting a Static IP Address

If you configured Static IP as the internet connection type:
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STEP 3 Click the System tab. STEP 4 Configure the following fields:
Static IP AddressEnter the static IP address of the phone. NetmaskEnter the subnet mask of the phone. GatewayEnter the IP address of the gateway.
For the SPA525G/525G2, you also have the following fields available: LAN MTULAN Maximum Transmission Unit size. Default value: 1500. Duplex ModeChoose one of the following to configure the speed/duplex for the phones Ethernet ports: Auto 10MBps/Duplex 10MBps/Half 100Mbps/Duplex 100MBps/Half
Configuring PPPoE Settings (Cisco SPA525G/525G2 only)

If you configured PPPoE as the internet connection type:
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the System tab.
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Setting Optional Network Parameters
5
Description Specifies the account name assigned by the ISP for connecting on a Point-to-Point Protocol over Ethernet (PPPoE) link. Specifies the password assigned by the ISP for connecting on a Point-to-Point Protocol over Ethernet (PPPoE) link. Specifies the service name assigned by the ISP for connecting on a Point-to-Point Protocol over Ethernet (PPPoE) link.
STEP 4 Configure the following fields:
Parameter PPPoE Login Name
PPPoE Login Password
PPPoE Service Name

STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the System tab. STEP 4 Configure the following fields: Parameter Host Name Domain Description The host name of the phone. The network domain of the phone. However, if using LDAP, see Configuring Lightweight Directory Access Protocol (LDAP) for the Cisco SPA300 Series and Cisco SPA500 Series IP Phones, page 85.
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5
Description DNS server used by the phone in addition to DHCP supplied DNS servers if DHCP is enabled; when DHCP is disabled, this is the primary DNS server. Defaults to 0.0.0.0. However, if using LDAP, see Configuring Lightweight Directory Access Protocol (LDAP) for the Cisco SPA300 Series and Cisco SPA500 Series IP Phones, page 85.
Parameter Primary DNS
Secondary DNS
DNS server used by the phone in addition to DHCP supplied DNS servers if DHCP is enabled; when DHCP is disabled, this is the secondary DNS server. Defaults to 0.0.0.0.
DNS Server Order
Specifies the method for selecting the DNS server. The options are Manual, Manual/DHCP, and DHCP/Manual. Do parallel or sequential DNS Query. With parallel DNS query mode, the phone sends the same request to all the DNS servers at the same time when doing a DNS lookup, the first incoming reply is accepted by the phone. Defaults to parallel. Not available on WIP310 or SPA525G/525G2.
DNS Query Mode
Syslog Server
Specify the syslog server name and port. This feature specifies the server for logging system information and critical events. If both Debug Server and Syslog Server are specified, Syslog messages are also logged to the Debug Server. The debug server name and port. This feature specifies the server for logging debug information. The level of detailed output depends on the debug level parameter setting. The debug level from 0-3. The higher the level, the more debug information is generated. Zero (0) means no debug information is generated. To log SIP messages, you must set the Debug Level to at least 2. Defaults to 0.
Debug Server
Debug Level
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Configuring VLAN Settings for IP Phones
5
Description Enables Network Time Protocol (NTP). Applies to the SPA525G only.
Parameter NTP Enable
Primary NTP Server
IP address or name of primary NTP server. The phones use these servers to synchronize its time. Defaults to blank. IP address or name of secondary NTP server. The phones use these servers to synchronize its time. Defaults to blank. Enable Bonjour networking that is used by Office Manager and Cisco Configuration Assistant to discover the Cisco IP phones. Choose yes to enable or no to disable.
Secondary NTP Server
Enable Bonjour (SPA525G/ 525G2 only)

If you use a VLAN your IP phone voice packets are tagged with the VLAN ID. (This section is not applicable to the WIP310.)
Configuring Cisco Discovery Protocol (CDP)

If you are using a Cisco switch, Cisco discovery protocol (CDP) is enabled. CDP is enabled by default. CDP is negotiation-based and determines which VLAN the IP phone resides in. CDP: Obtains the protocol addresses of neighboring devices and also discovers the platform of those devices. Shows information about the interfaces your router uses. Is media and protocol-independent.
If you are using a VLAN without CDP, you must enter a VLAN ID for the IP phone.
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Configuring Link Layer Discovery Protocol for Media Endpoint Devices (LLDP-MED)
The Cisco SPA500 Series and Cisco SPA300 Series IP Phones support LLDP-MED for deployment with Cisco or other third-party network connectivity devices that use a Layer 2 auto-discovery mechanism. Implementation of LLDP-MED is done in accordance with IEEE 802.1AB (LLDP) Specification of May 2005, and ANSI TIA1057 of April 2006. LLDP-MED is enabled by default. Cisco SPA IP Phones operates as LLDP-MED Media End Point Class III devices with direct LLDP-MED links to Network Connectivity Devices, according to the Media Endpoint Discovery Reference Model and Definition (ANSI TIA-1057 Section 6). The Cisco IP Phones support only the following limited set of TLVs as LLDP-MED Media Endpoint device class III: Chassis ID TLV Port ID TLV Time to live TLV Port Description TLV System Name TLV System Capabilities TLV IEEE 802.3 MAC/PHY Configuration/Status TLV (for wired network only) LLDP-MED Capabilities TLV LLDP-MED Network Policy TLV (for application type=Voice only) LLDP-MED Extended Power-Via-MDI TLV (for wired network only) LLDP-MED Firmware Revision TLV End of LLDPDU TLV
The outgoing LLDPDU contains all the above TLVs when if applicable. For the incoming LLDPDU, the LLDPDU is discarded if any of the following TLVs are missing. All other TLVs are not validated and ignored. Chassis ID TLV Port ID TLV
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Time to live TLV LLDP-MED Capabilities TLV LLDP-MED Network Policy TLV (for application type=Voice only) End of LLDPDU TLV
The phone sends out the shutdown LLDPDU if applicable. The LLDPDU frame contains the following TLVs: Chassis ID TLV Port ID TLV Time to live TLV End of LLDPDU TLV
There are some restrictions in the implementation of LLDP-MED on the Cisco IP Phones: Storage and retrieval of neighbor information is not supported. SNMP and corresponding MIBs are not supported. Recording and retrieval of statistical counters are not supported. There is no full validation of all TLVs; TLVs that do not apply to the phones are ignored. Protocol state machines as stated in the standards are only used for reference.
TLV Information
The following sections provide the TLV information. Chassis ID TLV For the outgoing LLDPDU, the TLV supports sub-type=5 (Network Address). When IP address is known, the value of Chassis ID is an octet of the INAN address family number followed by the octet string for the IPv4 address used for voice communication. If the IP address is unknown, the value for Chassis ID is 0.0.0.0. The only INAN address family supported is IPv4. Currently, IPv6 address for the Chassis ID is not supported. For the incoming LLDPDU, the Chassis ID is treated as an opaque value to form MSAP identifier. The value is not validated against its subtype. The Chassis ID TLV is mandatory as the first TLV. Only one Chassis ID TLV is allowed for the outgoing and incoming LLDPDUs.
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Port ID TLV For the outgoing LLDPDU, the TLV supports sub-type=3 (MAC address). The 6 octet MAC address for the Ethernet port is used for the value of Port ID in wired or wireless mode. For the incoming LLDPDU, the Port ID TLV is treated as an opaque value to form the MSAP identifier. The value is not validated against its sub-type. The Port ID TLV is mandatory as the second TLV. Only one Port ID TLV is allowed for the outgoing and incoming LLDPDUs. Time to Live TLV For the outgoing LLDPDU, the Time to live TTL value is 180 seconds. This is different from 120 seconds as recommended by the standard. For the shutdown LLDPDU, the TTL value is always 0. The Time to Live TLV is mandatory as the third TLV. Only one Time to Live TLV is allowed for the outgoing and incoming LLDPDUs. End of LLDPDU TLV The value is 2-octet, all zero. This TLV is mandatory and only one is allowed for the outgoing and incoming LLDPDUs. Port Description TLV For the outgoing LLDPDU, in the Port Description TLV, the value for the port description is the same as Port ID TLV for CDP. The incoming LLDPDU, the Port Description TLV, is ignored and not validated. Only one Port Description TLV is allowed for outgoing and incoming LLDPDUs. System Name TLV For the outgoing LLDPDU, in the System Name TLV, the value is the same as Platform TLV for CDP. For the Cisco SPA525G2, the name is SPA525G2. The incoming LLDPDU, the System Name TLV, is ignored and not validated. Only one System Name TLV is allowed for the outgoing and incoming LLDPDUs. System Capabilities TLV For the outgoing LLDPDU, in the System Capabilities TLV, the bit values for the 2 octet system capabilities field should be set for Bit 2 (Bridge) and Bit 5 (Phone) for a phone with a PC port. If the phone does not have a PC port, only Bit 5 should be set. The same system capability value should be set for the enabled capability field. For the incoming LLDPDU, the System Capabilities TLV is ignored. The TLV is not validated semantically against the MED device type. The System Capabilities TLV is mandatory for outgoing LLDPDUs. Only one System Capabilities TLV is allowed.
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IEEE 802.3 MAC/PHY Configuration/Status TLV The TLV is not for auto-negotiation, but for troubleshooting purposes. For the incoming LLDPDU, the TLV is ignored and not validated. For the outgoing LLDPDU, for the TLV, the octet value auto-negotiation support/status should be: Bit 0Set to 1 to indicate the auto-negotiation support feature is supported. Bit 1Set to 1 to indicate auto-negotiation status is enabled. Bit 2-7Set to 0.
The bit values for the 2 octets PMD auto-negotiation advertised capability field should be set to: Bit 1310BASE-T half duplex mode Bit 1410BASE-T full duplex mode Bit 11100BASE-TX half duplex mode Bit 10100BASE-TX full duplex mode Bit 15Unknown
Bit 10, 11, 13 and 14 should be set. The value for 2 octets operational MAU type should be set to reflect the real operational MAU type: 16100BASE-TX full duplex 15100BASE-TX half duplex 1110BASE-T full duplex 1010BASE-T half duplex
For example, in most cases, the phone is set to 100BASE-TX full duplex. The value 16 should then be set. The TLV is optional for a wired network and not applicable for a wireless network. The phone will send out this TLV only when in wired mode. When the phone is not set for auto-negotiation but specific speed/duplexity, for the outgoing LLDPDU TLV, bit 1 for the octet value auto-negotiation support/status should be clear (0) to indicate auto-negotiation is disabled. The 2 octets PMD autonegotiation advertised capability field should be set to 0x8000 to indicate unknown. The Cisco SPA525G/525G2 allows the administrator to set the switch operational mode to auto-negotiation or to a specific speed/duplexity.
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LLDP-MED Capabilities TLV For the outgoing LLDPDU, the TLV should have the device type 3 (End Point Class III) and with the following bits set for 2-octet Capability field:
Bit Position
Capability
0 1 4 5
LLDP-MED Capabilities Network Policy Extended Power via MDI-PD Inventory
For the incoming TLV, if the LLDP-MED TLV is not present, the LLDPDU is discarded. The LLDP-MED Capabilities TLV is mandatory and only one is allowed for the outgoing and incoming LLDPDUs. Any other LLDP-MED TLVs will be ignored if they present before the LLDP-MED Capabilities TLV. Network Policy TLV Outgoing LLDPDUThe phone will send out only one Network Policy TLV with the Application Type Value set to 1 (Voice). Before the VLAN or DSCP is determined, the Unknown Policy Flag (U) is set to 1. If the VLAN setting or DSCP is known, the value is set to 0. When the policy is unknown, all other values are set to 0. Before the VLAN is determined or used, the Tagged Flag (T) is set to 0. If the tagged VLAN (VLAN ID > 1) is used for the phone, the Tagged Flag (T) is set to 1. Reserved (X) is always set to 0. If the VLAN is used, the corresponding VLAN ID and L2 Priority will be set accordingly. VLAN ID valid value is range from 1-4094. However, VLAN ID=1 will never be used (limitation). If DSCP is used, the value range from 0-63 is set accordingly. Incoming LLDPDUMultiple Network Policy TLVs for different application types are allowed. The phone will only support and interpret the TLV with the application type=1 (Voice). TLVs for other application types are ignored and not validated. LLDP-MED Extended Power-Via-MDI TLV In the TLV for the outgoing LLDPDU, the binary value for Power Type is set to 0 1 to indicate the power type for phone is PD Device. The Power source for the phone is set PSE and local with binary value 1 1. The Power Priority is set to binary 0 0 0 0 to indicate unknown priority while the Power Value is set to maximum power value based on phone type:
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Phone Type
Power Value
SPA525G SPA50X SPA30X
125 120 100
For the incoming LLDPDU, the TLV is ignored and not validated. Only one TLV is allowed in the outgoing and incoming LLDPDUs. The phone will send out the TLV for wired network only.
NOTE The LLDP-MED standard was originally drafted in the context of Ethernet.
Discussion is ongoing for LLDP-MED for Wireless Networks. Refer to ANSI-TIA 1057, Annex C, C.3 Applicable TLV for VoWLAN, table 24. It is recommended that the TLV is not applicable in the context of the wireless network. This TLV is targeted for use in the context of PoE and Ethernet. The TLV, if added, will not provide any value for network management or power policy adjustment at the switch. LLDP-MED Inventory Management TLV This TLV is optional for Device Class III. For the outgoing LLDPDU, we support only Firmware Revision TLV. The value for the firmware revision is the firmware version. For the incoming LLDPDU, the TLVs are all ignored and not validated. Only one Firmware Revision TLV is allowed for the outgoing and incoming LLDPDUs. Final Network Policy Resolution and QoS For the Phone The following sections describe network policy and QoS for the IP phones. Special VLANs VLAN=0, VLAN=1 and VLAN=4095 are treated the same way as an untagged VLAN. As the VLAN is untagged, CoS is not applicable. Default QoS for SIP Mode If there is no network policy from CDP or LLDP-MED, the default network policy is used. CoS is based on configuration for the specific extension. It is applicable only if the manual VLAN is enabled and manual VLAN ID is not equal to 0, 1, or 4095. ToS is based on configuration for the specific extension.
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Default QoS for SPCP Mode If there is no network policy from CDP or LLDP-MED, the default network policy is used. CoS is based on a pre-defined value of 5. It is applicable only if the manual VLAN is enabled and manual VLAN ID is not equal to 0, 1, or 4095. ToS is based on precedence value from the StartMediaTransmission Message from the Unified Communications 500 Series for the Cisco SPA525G/525G2. However, ToS is based on the value specified for the specific extension in the web administration interface for the Cisco SPA50X IP phone. QoS Resolution for CDP If there is a valid network policy from CDP: If the VLAN=0, 1 or 4095, the VLAN will not be set, or the VLAN is untagged. CoS is not applicable, but DSCP is applicable. ToS is based on the default as previously described. If the VLAN > 1 and VLAN< 4095, the VLAN is set accordingly. CoS and ToS are based on the default as previously described. DSCP is applicable. For the Cisco SPA525G/525G2, when the VLAN is changed, the user sees the voice component refreshed when the IP address is changed. For the Cisco SPA50X, the phone reboots and restarts the fast start sequence.
QoS Resolution for LLDP-MED If CoS is applicable and if CoS=0, the default will be used for the specific extension as previously described. But the value shown on L2 Priority for TLV for outgoing LLDPDU is based on value used for extension 1. If CoS is applicable and if CoS != 0, CoS will be used for all extensions. If DSCP (mapped to ToS) is applicable and if DSCP=0, the default will be used for the specific extension as previously described. But the value show on DSCP for TLV for outgoing LLDPDU is based on value used for the extension 1. If DSCP is applicable and if DSCP != 0, DSCP will be used for all extensions. If the VLAN > 1 and VLAN < 4095, the VLAN is set accordingly. CoS and ToS are based on the default as previously described. DSCP is applicable. If there is a valid network policy for voice application from LLDP-MED PDU and if the tagged flag is set, the VLAN, L2 Priority (CoS) and DSCP (mapped to ToS) are all applicable. If there is a valid network policy for voice application from LLDP-MED PDU and if the tagged flag is not set, only the DPSC (mapped to ToS) is applicable.
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For the Cisco SPA525G/525G2, when the VLAN is changed, the user sees the voice component refreshed when IP address is changed. For the Cisco SPA50X, the phone reboots and restarts the fast start sequence. Co-Existence with CDP If both CDP and LLDP-MED are enabled, the network policy for the VLAN is determined by the last policy set or changed with either one of the discovery modes. If both LLDP-MED and CDP are enabled, during startup, the phone sends both CDP and LLDP-MED PDUs at the same time.
NOTE Inconsistent configuration and behavior for network connectivity devices for CDP
and LLDP-MED modes could result in an oscillating rebooting behavior for the phone due to switching to different VLANs. If the VLAN is not set via CDP and LLDP-MED, the VLAN ID that is configured manually is used. If the VLAN ID is not configured manually, no VLAN will be supported. DSCP is used and the network policy is determined by LLDP-MED if applicable. Wireless LAN Environments Network policy for the VLAN feature is not supported for wireless networks. The Wireless AP or Wireless router must be enabled for LLDP-MED as the Network Connectivity Device. The DSCP portion for network policy from Wireless AP/ Router will be supported if enabled. LLDP-MED and Multiple Network Devices If the same application type is used for network policy but different Layer 2 or Layer 3 QoS Network policies are received by the phones from multiple network connectivity devices, the last valid network policy is honored. To ensure deterministic and consistent of Network Policy, multiple network connectivity devices should not send out conflicting network policies for the same application type. LLDP-MED and IEEE 802.X The phones do not support IEEE 802.X and will not work in a 802.1X wired environment. However, IEEE 802.1X or Spanning Tree Protocols on network devices could result in delay of fast start response from switches.
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Configuring the VLAN Settings

STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the System tab. STEP 4 Under VLAN Settings, enter the following parameters:
Parameter Enable VLAN
Description Choose Yes to enable VLAN. Choose no to disable. If you use a VLAN without CDP (VLAN enabled and CDP disabled), enter a VLAN ID for the IP phone. Note that only voice packets are tagged with the VLAN ID. Do not use 1 for the VLAN ID. Choose No Limit, or 0-7 (default 0). The highest priority is 7. The priority applied to all frames, tagged and untagged. The phone modifies the frame priority only if the incoming frame priority is higher than this value. Enables VLAN and priority tagging on the phone data port (802.1p/q). This feature facilities tagging of the VLAN ID (802.1Q) and priority bits (802.1p) of the traffic coming from the PC port of the IP phone. Defaults to No. Choose Yes to enable the tagging algorithm.
VLAN ID
PC Port VLAN Highest Priority
Enable PC Port VLAN Tagging
PC Port VLAN ID
0-4095 (default 0). Value of the VLAN ID. The phone tags all the untagged frames coming from the PC (it will not tag frames with an existing tag). Enable CDP only if you are using a switch that has Cisco Discovery Protocol. CDP is negotiation based and determines which VLAN the IP phone resides in.
Enable CDP
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5
Description Choose yes to enable LLDP-MED for the phone to advertise itself to devices that use that discovery protocol. (By default, this setting is enabled.) When the LLDP-MED feature is enabled, after the phone has initialized and Layer 2 connectivity is established, the phone sends out LLDP-MED PDU frames. If the phone receives no acknowledgment, the manually configured VLAN or default VLAN will be used if applicable. If the CDP is used concurrently, the waiting period of 6 seconds is used. The waiting period will increase the overall startup time for the phone.
Parameter Enable LLDP-MED
Network Startup Delay
Setting this value causes a delay for the switch to get to the forwarding state before the phone will send out the first LLDP-MED packet. The default delay is 3 seconds. For configuration of some switches, you might need to increase this value to a higher value for LLDP-MED to work. Configuring a delay can be important for networks that use Spanning Tree Protocol.

The Cisco SPA525G/525G2 can be used in a virtual private network (VPN) to allow users secure access to the office phone network from remote locations. Users can connect their Cisco SPA525G/525G2 phones to the Internet and then use the VPN feature to securely access the company phone network. This feature works on the Cisco SPA525G/525G2 IP Phone using both SIP and SPCP.
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The Cisco SPA525G/525G2 works with the Cisco AnyConnect VPN client and the following VPN devices: Cisco 500 Series Secure Router Cisco 5500 Series Adaptive Security Appliance Cisco Unified Communications 520 Series
You must configure the SSL VPN device to ensure proper routing of voice data with desired VLAN and QoS at the end of the SSL VPN server. The following restrictions apply: HTTP proxy is not supported. SSL client certificate verification is not supported. CDP and VLAN tagging and QoS for the voice and PC port are not supported on the SSL VPN tunnel.
Because using the VPN requires internal phone resources, performance can suffer if using memory-intensive applications or configurations on the phone when the phone is connected using the VPN. The following restrictions apply: Only the G.711 Audio Codec is supported. SRTP for secured audio is not supported. Video monitoring is not supported.
To configure and use the Cisco SPA525G/525G2 on a VPN, you must do the following: 1. Configure the VPN on the VPN server using Cisco AnyConnect VPN client software. 2. Configure the VPN administrative settings on the Cisco SPA500 Series IP phone using the configuration utility. 3. Configure the VPN user settings using the configuration utility or on the IP phone using the phone menu.
Configuring the VPN on the Security Appliance

NOTE Specific configuration instructions are not presented in this document. For detailed
instructions for your particular device, see the application notes in the Cisco Small Business Support Community.
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STEP 1 Download the Cisco AnyConnect VPN client software from Cisco.com and install it
on the VPN server.

STEP 2 Download the Cisco IOS version that supports this feature and install it on the VPN
server.
STEP 3 Configure SSL VPN on the VPN server. STEP 4 Ensure the VPN is functional and you can connect to the VPN using the Cisco
AnyConnect VPN client.
Configuring the VPN on the Cisco SPA525G/525G2

SPCP Settings (Optional) If the phone will be connecting to the VPN using SPCP, configure these administrative settings using the configuration utility:
STEP 1 Log in to the configuration utility. STEP 2 Click the System tab. STEP 3 Under Optional Network Configuration, in the Alternate TFTP field, choose yes. STEP 4 In the TFTP Server field, enter the IP address of the Cisco Unified Communications
500 Series server. The phone obtains its software load from this server when the phone either boots in SPCP mode (if the Connect on Bootup field on the phone is set to yes), or connects to the VPN manually (by the user pressing Connect on the phone under the Network Configuration > VPN menu).
User Settings Then, enter the user settings for the phone, using either the configuration utility or the phone itself. To use the configuration utility:
STEP 1 :Log in to the configuration utility. STEP 2 Click Admin Login and advanced. (Not applicable to the SPA525G/525G2 in SPCP
mode.)
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STEP 3 Click the System tab. STEP 4 Under VPN Settings, enter the following:
In the VPN Server field, enter the IP address of the VPN server. In the VPN User Name and Password fields, enter the username and password to log in to the VPN server. These were created when you set up the VPN on the server. (Optional) Enter the VPN tunnel group, if required by your VPN server. (Optional) To connect to the VPN when the phone is powered on, in the Connect on Bootup field, choose yes.
STEP 5 Click Submit All Changes. If you did not choose yes in the Connect on Bootup field,
connect to the VPN on the phone by pressing the Setup button and choosing Network Configuration > VPN > Connect.
To use the phone interface:

STEP 1 On the phone, press the Setup button. STEP 2 Scroll to Network Configuration and press Select. STEP 3 Scroll to Web Server and ensure that it is enabled. Press the right arrow key if it is
not enabled.
STEP 4 Scroll to VPN and press the right arrow key. STEP 5 Under VPN server, enter the IP address of the VPN server. STEP 6 Enter the username to log in to the VPN server. STEP 7 Enter the password for the user. STEP 8 (Optional) Enter the tunnel group, if required by the VPN server. STEP 9 (Optional) To connect to the VPN when the phone is powered on, ensure that
Connect on Bootup is enabled.

STEP 10 To connect to the VPN, ensure that Connect is enabled. STEP 11 Press Save. After the VPN connection is established, a VPN icon appears in the
upper right of the phone display screen.
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To view the VPN status, either: Use the configuration utility: Click Admin Login and advanced. (Not applicable to the SPA525G/ 525G2 in SPCP mode.) Click the Info tab.
Use the phone menu: Press the Setup button. Scroll to Status and press Select. Scroll to VPN Status and press Select.
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Provisioning Basics
The Provisioning Tab and its fields are for service provider use only and are not needed in non-SP deployments. This chapter discusses: Provisioning Capabilities, page 161 IP Phone Configuration Profiles, page 163 Sample Configuration File, page 165 Upgrading, Resyncing, and Rebooting Phones, page 166 Redundant Provisioning Servers, page 169 Retail Provisioning, page 169 Automatic In-House Preprovisioning, page 170 Configuration Access Control, page 171 Using HTTPS, page 172
VARs and service providers should refer to other documentation, depending on your configuration:
Cisco Small Business IP Telephony Devices Provisioning Guide (service

provider login required) Cisco SPA3xx, SPA50xG, and SPA525G SPC Templates for Configuration Files, available on the Cisco Support Community at: https://supportforums.cisco.com/docs/DOC-10008
Cisco SPA 9000 Voice System Administration Guide

Service provider documentation
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The Cisco IP phones provide for secure provisioning and remote upgrade. Provisioning is achieved through configuration profiles transferred to the device via TFTP, HTTP, or HTTPS. The Cisco IP phones can be configured to automatically resync their 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 Cisco IP phones accept profiles in XML format, or alternatively in a proprietary binary format, which is generated by a profile compiler tool, SIP Profile Compiler (SPC), available from Cisco. The Cisco IP phones support up to 256-bit symmetric key encryption of profiles. For the initial transfer of the profile encryption key (initial provisioning stage), the Cisco IP phones can receive a profile from an encrypted channel (HTTPS with client authentication), or can resync to a binary profile generated by the Cisco SIP profile compiler. In the latter case, the SIP profile compiler can encrypt the profile specifically for the target Cisco IP phones, without requiring an explicit key exchange. Remote firmware upgrade is achieved via TFTP or HTTP or HTTPS (TFTP or HTTP for WIP310). Remote upgrades are controlled by configuring the desired firmware image URL into the Cisco IP phone via a remote profile resync.
Provisioning Configuration from Phone Keypad

Remote provisioning can be performed from a phone keypad. After the user enters the IP address of the provisioning server, the unit resyncs to a known path name. This feature enables service providers to allow VARs to install and provision Cisco phones.
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To provision from the phone: Cisco SPA303 and Cisco SPA50XG:
STEP 1 Press Setup, then scroll to Profile Rule. STEP 2 Enter the profile rule using the following format, then press the Resync soft button.
protocol://server[:port]/profile_pathname For example: tftp://192.168.1.5/spa504.cfg
If no protocol is specified, TFTP is assumed. If no server-name is specified, the host that requests the URL is used as server-name. If no port is specified, the default port is used (69 for TFTP, 80 for HTTP, and 443 for HTTPS). then the address can be entered in and press Resync. The status of the remote customization process is shown by the phones mute button blinking in the following patterns: Red/orange slow blink (1.0 seconds on, 1.0 seconds off): Contacting server, server not resolvable, not reachable, or down Red/orange slow blink (0.2 seconds on, 0.2 seconds off, 0.2 seconds on, 1.4 seconds off): Server responded with file not found or corrupt file
WIP310
STEP 1 Press the Select button to choose Settings and press the Select button again. STEP 2 Navigate to Misc Settings. STEP 3 Navigate to profile rule. Enter the profile rule in the following format:
protocol://server[:port]/profile_pathname
For example, to have the WIP310 provisioning done by the Cisco SPA 9000 Voice System, enter: 192.168.2.64/cfg/generic.xml
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SPA525G/525G2
STEP 1 Press the Setup button. STEP 2 Scroll to Device Administration and press Select. STEP 3 Scroll to Profile Rule and press Select. STEP 4 Enter the profile rule using the following format, then press the Resync soft button.
protocol://server[:port]/profile_pathname For example: tftp://192.168.1.5/spa525.cfg

The IP phone configuration profile defines the parameter values for a specific device. The configuration profile can be used in two formats: Open (XML-style) format Proprietary, plain-text format
The XML-style format lets you use standard tools to compile the parameters and values. To protect confidential information contained in the configuration profile, this type of file is generally delivered from the provisioning server to the IP phone over a secure channel provided by HTTPS. 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. The following is an example of a valid XML profile: <flat-profile> <Admin_Passwd>some secret</Admin_Passwd> <Upgrade_Enable>Yes</Upgrade_Enable> </flat-profile>
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The names of parameters in XML profiles can generally be inferred from the Cisco IP phones configuration web pages, by substituting underscores (_) for spaces and other control characters. Further, 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. The plain-text configuration file uses a proprietary format, which can be encrypted to prevent unauthorized use of confidential information. By convention, the profile is named with the extension .cfg (for example, spa504.cfg). The Cisco SIP Profile Compiler (SPC) tool is provided for compiling the plain-text file containing parameter-value pairs into an encrypted CFG file. The SPC tool is available from Cisco for the Win32 environment (spc.exe) and Linux-i386-elf environment (spclinux-i386-static). Availability of the SPC tool for the OpenBSD environment is available on a case-by-case basis. 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. The following is an example of a valid text source profile for input to the SPC tool: Admin_Passwd some secret; Upgrade_Enable Yes; Parameters in the case of source text files for the SPC tool are similarly named, except that to differentiate Line 1, 2, 3, and 4, the appended strings ([1], [2], [3], or [4]) are used. For example, the Line 1 Proxy is named Proxy[1] in source text profiles for input to the SPC.
Obtaining the SPC Tool

The SPC tool is available on cisco.com. To obtain the software:
STEP 1 Go to:
http://tools.cisco.com/support/downloads/go/Redirect.x?mdfid=282414147
STEP 2 Click the Profile Compiler (SPC) Tool link. STEP 3 Choose the version of software that is installed on the phones. STEP 4 Follow the links to download the software.
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Sample Configuration File
6
General Purpose Parameters
These are configured in the General Purpose Parameters section of the Provisioning tab. These parameters can be used as variables in provisioning and upgrade rules. They are referenced by prepending the variable name with a $ character, such as $GPP_A. You can optionally Require Admin Password to Reset Unit to Factory Defaults (see last line of sample config file).
Sample Configuration File

Following is a sample configuration file:
Set_Local_Date_(mm/dd) "" ; Set_Local_Time_(HH/mm) "" ; Time_Zone "GMT-07:00" ; # options: GMT-12:00/GMT-11:00/GMT-10:00/GMT-09:00/ GMT-08:00/GMT-07:00/GMT-06:00/GMT-05:00/GMT-04:00/GMT-03:30/GMT-03:00/GMT02:00/GMT-01:00/GMT/GMT+01:00/GMT+02:00/GMT+03:00/GMT+03:30/GMT+04:00/ GMT+05:00/GMT+05:30/GMT+05:45/GMT+06:00/GMT+06:30/GMT+07:00/GMT+08:00/ GMT+09:00/GMT+09:30/GMT+10:00/GMT+11:00/GMT+12:00/GMT+13:00 FXS_Port_Impedance "600" ; # options: 600/900/600+2.16uF/900+2.16uF/ 270+750||150nF/220+820||120nF/220+820||115nF/370+620||310nF FXS_Port_Input_Gain "-3" ; FXS_Port_Output_Gain "-3" ; DTMF_Playback_Level "-16" ; DTMF_Playback_Length ".1" ; Detect_ABCD "Yes" ; Playback_ABCD "Yes" ; Caller_ID_Method "Bellcore(N.Amer,China)" ; # options: Bellcore(N.Amer,China)/DTMF(Finland,Sweden)/DTMF(Denmark)/ETSI DTMF/ETSI DTMF With PR/ETSI DTMF After Ring/ETSI FSK/ETSI FSK With PR(UK) FXS_Port_Power_Limit "3" ; # options: 1/2/3/4/5/6/7/8 Protect_IVR_FactoryReset No ;
NOTE You can optionally require an admin password to reset the phone to factory
defaults by setting the last line parameter to yes. If you are a service provider with a password, see the Cisco Small Business IP
Telephony Devices Provisioning Guide.
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Upgrading, Resyncing, and Rebooting Phones

Cisco IP phones support secure remote provisioning and firmware upgrades. You can generate configuration profiles using common, open-source tools that integrate with service provider provisioning systems. Supported transport protocols include TFTP, HTTP, and HTTPS with client certificates. 256-bit symmetric key encryption of profiles is supported. In addition, an unprovisioned Cisco IP phone can receive an encrypted profile specifically targeted for that device without requiring an explicit key. a secure first-time provisioning mechanism using SSL functionality. 1User intervention is not required to initiate or complete a profile update or firmware upgrade. The Cisco IP phone upgrade logic is capable of automating multi-stage upgrades, if intermediate upgrades are required to reach a future upgrade state from an older release. A profile resync is only attempted when the Cisco IP phone is idle, because this may trigger a software reboot. General purpose parameters are provided to help service providers manage the provisioning process. Each Cisco IP phone can be configured to periodically contact a normal provisioning server (NPS). Communication with the NPS does not require the use of a secure protocol because the updated profile is encrypted by a shared secret key. The NPS can be a standard TFTP, HTTP or HTTPS server. The administrator can upgrade, reboot, restart, or resync Cisco IP phones using the configuration utility. The administrator can also perform these tasks using a SIP notify message and bypassing the configuration utility.
Upgrading Firmware on a Phone

Use the upgrade URL to upgrade firmware on the Cisco IP phone. You can upgrade from either a TFTP or HTTP server. The Upgrade Enable parameter on the Provisioning web page must be set to yes.
STEP 1 Log in to the configuration utility. STEP 2 Click the Provisioning tab. STEP 3 Under Firmware Upgrade, in the Upgrade Enable field, choose yes.
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Enter the upgrade command into your browsers address bar to upgrade firmware on a phone. Use the following syntax: http://phone-ip-address/admin/upgrade?protocol://servername[:port]]/firmware-path Protocol defaults to TFTP. Server name is the host requesting the URL. Port is the port of the protocol being used (for example, 69 for TFTP or 80 for HTTP). Firmware-path defaults to /spa.bin (for example, http://192.168.2.217/ admin/upgrade?tftp://192.168.2.251/spa.bin) for SPA phones and / wip310.img for the WIP310. The firmware-pathname is typically the file name of the binary located in a directory on the TFTP or HTTP server.
Firmware Upgrade Parameters

The following table defines the function and usage of each parameter in the Firmware Upgrade section of the Provisioning tab.
Parameter Upgrade_Enable
Description Enables firmware upgrade operations independently of resync actions. Defaults to Yes.
Upgrade_Error_Retry_Delay
The upgrade retry interval (in seconds) applied in case of upgrade failure. The device has a firmware upgrade error timer that activates after a failed firmware upgrade attempt. The timer is initialized with the value in this parameter. The next firmware upgrade attempt occurs when this timer counts down to zero. The default is 3600 seconds.
Downgrade_Rev_Limit
Enforces a lower limit on the acceptable version number during a firmware upgrade or downgrade. The device does not complete a firmware upgrade operation unless the firmware version is greater than or equal to this parameter. The default is (empty).
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Description This parameter is a firmware upgrade script with the same syntax as Profile_Rule. Defines upgrade conditions and associated firmware URLs. The default is (empty).
Parameter Upgrade_Rule
Log_Upgrade_Request_Msg
Syslog message issued at the start of a firmware upgrade attempt. The default is
$PN $MAC -- Requesting upgrade $SCHEME:// $SERVIP:$PORT$PATH
Log_Upgrade_Success_Msg
Syslog message issued after a firmware upgrade attempt completes successfully. The default is
$PN $MAC -- Successful upgrade $SCHEME:// $SERVIP:$PORT$PATH -- $ERR
Log_Upgrade_Failure_Msg
Syslog message issued after a failed firmware upgrade attempt. The default is $PN $MAC -- Upgrade failed: $ERR.
License Keys
This field is empty.
Resyncing a Phone
You can resync an IP phone to a specific remote profile. The configuration of the phone you resync will match the configuration of the remote phone. The phone can be configured to resync its internal configuration state to a remote profile periodically and on power up.
NOTE The phone resyncs only when it is idle.
Use the following syntax to resync a phones profile to a profile on a TFTP, HTTP, or HTTPS server: http://phone-ip-addr/admin/resync?protocol://servername[:port]/profile-pathname Parameter following resync? defaults to the Profile Rule setting on the web server Provisioning page. Protocol defaults to TFTP.
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Provisioning Basics
Redundant Provisioning Servers
6
Server-name defaults to the host requesting the URL. Port defaults to: 69 for TFTP 80 for HTTP 443 for HTTPS
Profile-path defaults to the path to the new resync profile (for example, http://192.168.2.217admin/resync?tftp://192.168.2.251/spaconf.cfg).
Rebooting a Phone
You can remotely reboot a Cisco IP phone if needed. Use the following syntax to reboot a phone: http://phone-ip-address/admin/reboot
Redundant Provisioning Servers

The provisioning server may be specified as an IP address or as a fully qualified domain name (FQDN). The use of a FQDN facilitates the deployment of redundant provisioning servers. When the provisioning server is identified through a FQDN, the Cisco IP phone attempts to resolve the FQDN to an IP address through DNS. Only DNS A-records are supported for provisioning; DNS SRV address resolution is not available for provisioning. The Cisco IP phone continues to process Arecords until the first server responds. If no server associated with the A-records responds, the Cisco IP phone logs an error to the syslog server.
Retail Provisioning
The Cisco IP phone includes the web-based configuration utility that displays internal configuration and accepts new configuration parameter values. The server also accepts a special URL command syntax for performing remote profile resync and firmware upgrade operations.
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Provisioning Basics
Automatic In-House Preprovisioning
In a retail distribution model, a customer purchases a Cisco voice endpoint device, and subsequently subscribes to a particular service. The customer first signs on to the service and establishes a VoIP account, possibly through an online portal. Subsequently, the customer binds the particular device to the assigned service account. To do so, the unprovisioned Cisco IP phone is instructed to resync with a specific provisioning server through a resync URL command. The URL command typically includes an account PIN number or alphanumeric code to associate the device with the new account. In the following example, a device at the DHCP-assigned IP address 192.168.1.102 is instructed to provision itself to the SuperVoIP service:
http://192.168.1.102/admin/resync?https://prov.supervoip.com/cisco-init/ 1234abcd
In this example, 1234abcd is the PIN number of the new account. The remote provisioning server is configured to associate the Cisco IP phone that is performing the resync request with the new account, based on the URL and the supplied PIN. Through this initial resync operation, the Cisco IP phone is configured in a single step, and is automatically directed to resync thereafter to a permanent URL on the server. For example:
https://prov.supervoip.com/cisco-init
For both initial and permanent access, the provisioning server relies on the Cisco IP phone client certificate for authentication and supplies correct configuration parameter values based on the associated service account.1-5
Automatic In-House Preprovisioning

Using the configuration utility and issuing a resync URL is convenient for a customer in the retail deployment model, but it is not as convenient for preprovisioning a large number of units. The Cisco IP phone supports a more convenient mechanism for in-house preprovisioning. With the factory default configuration, a Cisco IP phone automatically tries to resync to a specific file on a TFTP server, whose IP address is offered as one of the DHCP-provided parameters. This lets a service provider connect each new Cisco IP phone to a LAN environment configured to
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Provisioning Basics
Configuration Access Control
preprovision phones. Any new Cisco IP phone connected to this LAN automatically resyncs to the local TFTP server, initializing its internal state in preparation for deployment. Among other parameters, this preprovisioning step configures the URL of the Cisco IP phone provisioning server. Subsequently, when a new customer signs up for service, the preprovisioned Cisco IP phone can be simply bar-code scanned, to record its MAC address or serial number, before being shipped to the customer. Upon receiving the unit, the customer connects the unit to the broadband link. On power-up the Cisco IP phone already knows the server to contact for its periodic resync update.
Configuration Access Control

Besides configuration parameters that control resync and upgrade behavior, the Cisco IP phone provides mechanisms for restricting end-user access to various parameters. The Cisco IP phone firmware provides specific privileges for login to a User account and an Admin account. The Admin account is designed to give the service provider or VAR configuration access to the Cisco IP phone, while the User account is designed to give limited and configurable control to the end user of the device. The User and Admin accounts can be independently password protected. The configuration parameters available to the User account are completely configurable in the Cisco IP phone, on a parameter-by-parameter basis. Optionally, user access to the Cisco IP phone configuration utility can be totally disabled. The Internet domains accessed by the Cisco IP phone for resync, upgrades, and SIP registration for Line 1 can be restricted.
Restricting User Access to the Phone Interface Menus (Cisco SPA300 and Cisco SPA500 Series)
You can restrict the menus and options that phone users see when they use the phone interface. The Phone-UI-user-mode parameter can be enabled in a provisioning file or in the configuration utility. Specific parameters are then designated as na or ro. Parameters designated as na will not appear on the users phone interface. Parameters designated as ro will not be editable by the user.
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Provisioning Basics
Using HTTPS
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You can set the Phone-UI-user-mode parameter in the configuration utility, but you then need to designate individual parameters as na or ro using provisioning files. For more information on provisioning, see the Cisco Small Business IP Telephony Devices Provisioning Guide on cisco.com. To change the Phone-UI-user-mode parameter using the web administration interface:
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the System tab. STEP 4 In the System Configuration section, in the Phone-UI-user-mode field, choose yes. STEP 5 Click Submit All Changes.
Using HTTPS
The Cisco IP phone provides a reliable and secure provisioning strategy based on HTTPS requests from the Cisco IP phone to the provisioning server, using both server and client certificates for authenticating the client to the server and the server to the client. To use HTTPS with Cisco IP phones, you must generate a Certificate Signing Request (CSR) and submit it to Cisco. The Cisco IP phone generates a certificate for installation on the provisioning server that is accepted by Cisco IP phones when they seek to establish an HTTPS connection with the provisioning server. The Cisco IP phone implements up to 256-bit symmetric encryption, using the American Encryption Standard (AES), in addition to 128-bit RC4. The Cisco IP phone supports the Rivest, Shamir, and Adelman (RSA) algorithm for public/ private key cryptography.
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Provisioning Basics
Using HTTPS
6
Server Certificates
Each secure provisioning server is issued an secure sockets layer (SSL) server certificate, directly signed by Cisco. The firmware running on the Cisco IP phone clients recognizes only these certificates as valid. The clients try to authenticate the server certificate when connecting via HTTPS, and reject any server certificate not signed by Cisco. This mechanism protects the service provider from unauthorized access to the Cisco IP phone endpoint, or any attempt to spoof the provisioning server. This might allow the attacker to reprovision the Cisco IP phone to gain configuration information, or to use a different VoIP service. Without the private key corresponding to a valid server certificate, the attacker is unable to establish communication with a Cisco IP phone.
Client Certificates
In addition to a direct attack on the Cisco IP phone, an attacker might attempt to contact a provisioning server using a standard web browser, or other HTTPS client, to obtain the Cisco IP phone configuration profile from the provisioning server. To prevent this kind of attack, each Cisco IP phone also carries a unique client certificate, also signed by Cisco, including identifying information about each individual endpoint. A certificate authority root certificate capable of authenticating the device client certificate is given to each service provider. This authentication path allows the provisioning server to reject unauthorized requests for configuration profiles.
Obtaining a Server Certificate

To obtain a server certificate:
STEP 1 Contact a Cisco support person who will work with you on the certificate process.
If you are not working with a specific support person, you can email your request to [email protected].)
STEP 2 Generate a private key that will be used in a CSR (Certificate Signing Request).
This key is private and you do not need to provide this key to Cisco support. Use open source "openssl" to generate the key. For example: openssl genrsa -out <file.key> 1024
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Using HTTPS
6
For example: openssl req -new -key <file.key> -out <file.csr> You must have the following information: Subject fieldEnter the Common Name (CN) that must be a FQDN (Fully Qualified Domain Name) syntax. During SSL authentication handshake, the SPA 9000 verifies that the certificate it receives is from the machine that presented it. Server's hostnameFor example, provserv.domain.com. Email addressEnter an email address so that customer support can contact you if needed. This email address is visible in the CSR.
STEP 3 Generate CSR a that contains fields that identify your organization, and location.
STEP 4 Email the CSR (in zip file format) to the Cisco support person or to linksys-
[email protected]. The certificate is signed by Cisco and given to you.
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7
Configuring Regional Parameters and Supplementary Services
Use the Regional tab to configure regional and local settings, such as Vertical Service Activation codes (star codes), Vertical Service Announcement Codes, and local language and dictionary. See the following sections: Advanced Scripting for Cadences, Call Progress Tones, and Ring Tones, page 176 Call Progress Tones, page 179 Distinctive Ring Patterns, page 179 Control Timer Values (sec), page 180 Vertical Service Announcement Codes (Cisco SPA300 and Cisco SPA500 Series), page 186 Miscellaneous Parameters, page 189 Localizing Your IP Phone, page 190 Selecting a Display Language, page 193
Cisco IP phones have configurable call progress tones. Parameters for each type of tone can include number of frequency components, frequency and amplitude of each component, and cadence information. The call progress tone pass-through feature lets you hear call progress tones (such as ringing) that are generated from the far-end network.
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Advanced information on defining tones an and cadences follows. A CadScript is a mini-script that specifies the cadence parameters of a signal. It can be up to 127 characters. The syntax follows: 1[;S2]where Si=Di(oni,1/offi,1[,oni,2/offi,2[,oni,3/offi,3[,oni,4/offi,4[,oni,5/ offi,5[,oni,6/offi,6]]]]]) and is known as a section, oni,j and offi,j are the on/off duration in seconds of a segment and i = 1 or 2, and j = 1 to 6. Di is the total duration of the section in seconds. All durations can have up to 3 decimal places to provide 1 ms resolution. The wildcard character * stands for infinite duration. The segments within a section are played in order and repeated until the total duration is played.
Example 1: Normal Ring

60(2/4) Number of Cadence Sections = 1 Cadence Section 1: Section Length = 60 s Number of Segments = 1 Segment 1: On=2s, Off=4s Total Ring Length = 60s
Example 2: Distinctive Ring (short,short,short,long)

60(.2/.2,.2/.2,.2/.2,1/4) Number of Cadence Sections = 1 Cadence Section 1: Section Length = 60s Number of Segments = 4 Segment 1: On=0.2s, Off=0.2s Segment 2: On=0.2s, Off=0.2s Segment 3: On=0.2s, Off=0.2s Segment 4: On=1.0s, Off=4.0s Total Ring Length=60s
A ToneScript is a mini-script that specifies the frequency, level and cadence parameters of a call progress tone. It can contain up to 127 characters. The syntax follows: FreqScript;Z1[;Z2]. The section Zi is similar to the Si section in a CadScript except that each on/off segment is followed by a frequency components parameter: Zi = Di(oni,1/offi,1/fi,1[,oni,2/offi,2/fi,2 [,oni,3/offi,3/fi,3 [,oni,4/ offi,4/fi,4 [,oni,5/offi,5/fi,5 [,oni,6/offi,6/fi,6]]]]])
Cisco Small Business SPA300 Series, SPA500 Series, and WIP310 IP Phone Administration Guide 176

where fi,j = n1[+n2]+n3[+n4[+n5[+n6]]]]] and 1 < nk < 6 indicates which of the frequency components given in the FreqScript are used in that segment; if more than one frequency component is used in a segment, the components are summed together.
Example 1: Dial Tone

350@-19,440@-19;10(*/0/1+2) Number of Frequencies = 2 Frequency 1 = 350 Hz at 19 dBm Frequency 2 = 440 Hz at 19 dBm Number of Cadence Sections = 1 Cadence Section 1: Section Length = 10 s Number of Segments = 1 Segment 1: On=forever, with Frequencies 1 and 2 Total Tone Length = 10s Example 2: Stutter Tone 350@-19,440@-19;2(.1/.1/1+2);10(*/0/1+2) Number of Frequencies = 2 Frequency 1 = 350 Hz at 19 dBm Frequency 2 = 440 Hz at 19 dBm Number of Cadence Sections = 2 Cadence Section 1: Section Length = 2s Number of Segments = 1 Segment 1: On=0.1s, Off=0.1s with Frequencies 1 and 2 Cadence Section 2: Section Length = 10s Number of Segments = 1 Segment 1: On=forever, with Frequencies 1 and 2 Total Tone Length = 12s
Example 3: SIT Tone

985@-16,1428@-16,1777@-16;20(.380/0/1,.380/0/2,.380/0/3,0/4/0) Number of Frequencies = 3 Frequency 1 = 985 Hz at 16 dBm Frequency 2 = 1428 Hz at 16 dBm Frequency 3 = 1777 Hz at 16 dBm Number of Cadence Sections = 1 Cadence Section 1: Section Length = 20s Number of Segments = 4 Segment 1: On=0.38s, Off=0s, with Frequency 1 Segment 2: On=0.38s, Off=0s, with Frequency 2 Segment 3: On=0.38s, Off=0s, with Frequency 3 Segment 4: On=0s, Off=4s, with no frequency components Total Tone Length = 20s
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A RingScript is a mini-script that describes a ring tone. The syntax follows: n=ring-tone-name;w=waveform-id-or-path;c=cadence-id;b=breaktime;t=total-time ring-tone-name is a name to identify this ring tone specification. This name will appear on the Ring Tone menu of the phone. The same name can be used in a SIP Alert-Info header in an inbound INVITE request to tell the phone to play the corresponding ring tone specification. Because of this, the name should contain characters allowed in a URL only. Waveform-id is the index of the desired waveform to use in this ring tone specification. There are 4 built-in waveforms: 1 = A classic phone with mechanical bell 2 = Typical phone ring 3 = A classic ring tone 4 = A wide-band frequency sweep signal
This field can also be a network path (url) to download a ring tone data file from a server on-the-fly. In this case, the syntax of the field is w=[tftp://]hostname[:port]/path. cadence-id is the index of the desired cadence to play the given waveform. 8 cadences (18) as defined in <Cadence 1> through <Cadence 8>. Cadence-id can be 0 If w=3,4, or an url. Setting c=0 implies the on-time is the natural length of the ring tone file. break-time specifies the number of seconds to break between two bursts of ring tone, such as b=2.5 total-time specifies the total number of seconds to play the ring tone before it times out
Example 1: SIT Tone

n=Classic-1,w=3;c=1 n=Simple-1,w=2;c=1
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Call Progress Tones
Call Progress Tones

For definitions of all call progress tones, see Call Progress Tones section on page 255.
Distinctive Ring Patterns

Ring cadence defines the ringing pattern that announces a telephone call.
NOTE The WIP310 has only eight distinctive ring pattern fields.
The pattern is:

length(on/off)
where: Length: The total length of the ring On: The number of on seconds Off: The number of off seconds.
Example 1: Normal Ring

60(2/4)
Number of Cadence Sections = 1 Cadence Section 1: Section Length = 60 s Number of Segments = 1 Segment 1: On=2s, Off=4s Total Ring Length = 60s
Example 2: Distinctive Ring (short,short,short,long)

60(.2/.2,.2/.2,.2/.2,1/4)
Number of Cadence Sections = 1 Cadence Section 1: Section Length = 60s Number of Segments = 4 Segment 1: On=0.2s, Off=0.2s
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Control Timer Values (sec)
Segment 2: On=0.2s, Off=0.2s Segment 3: On=0.2s, Off=0.2s Segment 4: On=1.0s, Off=4.0s Total Ring Length=60s

The following table describes all control timer parameters. Each value is displayed in seconds.
Field Reorder Delay
Description Delay after far end hangs up before reorder (busy) tone is played. 0 = plays immediately, inf = never plays. Range: 0255 seconds. Set to 255 to return the phone immediately to on-hook status and to not play the tone. Defaults to 5.
Call Back Expires
Expiration time in seconds of a call back activation. Range: 0 65535 seconds. Defaults to 1800.
Call Back Retry Intvl Call Back Delay
Call back retry interval in seconds. Range: 0255 seconds. Defaults to 30. Delay after receiving the first SIP 18x response before declaring the remote end is ringing. If a busy response is received during this time, the Cisco IP phone still considers the call as failed and keeps on retrying. Defaults to 0.5.
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Configuring Supplementary Services (Star Codes)
Field Interdigit Long Timer
Description 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: 064 seconds. Setting this value high can result in a longer post dialing delay (PDD), which is the time between the start of a call and the time the phone starts ringing. A value that is too low can result in dialed digits not being correctly recognized. Defaults to 10.
Interdigit Short Timer
Short timeout between entering digits when dialing. The Interdigit Short Timer is used after any one digit, if at least one matching sequence is complete as dialed, but more dialed digits would match other as yet incomplete sequences. Range: 064 seconds. Defaults to 3.

The Cisco IP phones provide native support of a large set of enhanced or supplementary services (also known as star codes). A user can enter star codes (such as *21 for call forward, followed by the target number) to perform call features such as call return, blind call transfers, call pickup, and so on. These codes can be handled locally by the phone or to be sent to the network as an INVITE to the service provider.
NOTE Some service providers choose to disable star codes. See Configuring
Supplementary Services (Star Codes) section on page 181.
Entering Star Code Values

The phone provides default values for star codes. You can change the default star code values for your area or region.
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To configure star code values:

STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the Regional tab. STEP 4 Under Vertical Service Activation Codes, enter the values you want to change for
the codes.
The codes are as follows: Call Return (*69)Calls the last caller, regardless which extension. Blind Transfer (*98)Allows the user to transfer a call to another number without waiting for the other party to pick up. Call Back Act (*66)Periodically redials the last busy number (every 30 seconds by default) until it rings or until the attempt expires (30 min by default), regardless which extension. Only one call back operation can be ordered at a time. A new order automatically cancels the previous order. Call Back Deact (*86)Cancels the last call back operation. Call Forward All Act (*72)Call forwards all inbound calls. Applies to primary extension only. Call Forward All Deact (*73)Cancels call forward all. Applies to primary extension only.0 Call Forward Busy Act (*90)Call forwards on busy. Applies to primary extension only. Call Forward Busy Deact (*91)Cancels call forward on busy. Applies to primary extension only. Call Forward No Answer Act (*92)Call forwards if no answer. Applies to primary extension only. Call Forward No Answer Deact (*93)Cancels call forward no answer. Applies to primary extension only.
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CW Act (*56)Enables call waiting. For example, if call waiting is turned off globally, this star code will turn on call waiting until the CW Deact code is entered. CW Deact (*57)Deactivates call waiting. For example, if call waiting is turned on globally, this star code deactivates call waiting until the CW Act code is entered. CW Per Call Act (*71)Enables call waiting for a single call. For example, if call waiting is turned off globally, this star code will turn on call waiting for that call. CW Per Call Deact (*70)Deactivates call waiting for a single call. For example, if call waiting is turned on globally, this star code deactivates call waiting for that call. Block CID Act (*67)Blocks caller ID on all outbound calls. Applies to all extensions. Block CID Deact (*68)Deactivates caller ID blocking on outbound calls. Applies to all extensions. Block CID Per Call (*81)Blocks caller ID on the next outbound call (on the current call appearance only). Block CID Per Call Deact (*82)Deactivates caller ID blocking on the next outbound call (on the current call appearance only). Block ANC ActBlocks anonymous calls. Applies to all extensions. Block ANC DeactDeactivates anonymous call blocking. Applies to all extensions. DND Act (*78)Activates Do Not Disturb. Applies to all extensions. DND Deact (*79)Deactivates Do Not Disturb. Applies to all extensions. Secure All Call Act (*16)Defaults to prefer to use encrypted media (voice codecs). Secure No Call Act (*17)Defaults to prefer to use unencrypted media for all outbound calls. Applies to all extensions. Secure One Call Act (*18)Prefers to use encrypted media for the outbound call (on this call appearance only). Secure One Call Deact (*19)Prefers to use unencrypted media for the outbound call (on this call appearance only).
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Paging (*96)Pages the number called. Call Park (*38)Parks a call on an entered line number. Call UnPark Code (*39)Retrieves a call from an entered line number. Call Pickup (*36)Picks up a call at an entered extension. Group Call Pickup (*37)Picks up a ringing call at a group of extensions. Media Loopback Code (*03)A service provider can set up a test call from an IP media loopback server (the source) to a subscribers VoIP device (the mirror). The test call provides statistical reporting on network performance and audio quality. Depending on the sources capabilities, the SP can see packet jitter, loss, and delay (although Media Loopback cannot identify an offending hop). This helps the SP identify an offending hop that could be causing issues in VoIP calls to a subscriber. The test results can also provide audio quality scoring, which lets a SP better understand the subscribers experience.
Referral Services CodesOne or more * codes can be configured into this parameter, such as *98, or *97|*98|*123, and so on. The maximum total length is 79 characters. 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 Cisco IP phone to perform a blind transfer to a target number that is prepended by the service * code. For example:
a. After the user dials *98, the Cisco IP phone plays a special prompt tone while waiting for the user the enter a target number (which is validated according to the dial plan as in normal dialing). b. When a complete number is entered, the Cisco IP phone sends a blind REFER to the holding party with the Refer-To target equals to *98 target_number. This feature allows the Cisco IP phone to hand off a call to an application server to perform further processing, such as call park. The * codes should not conflict with any of the other vertical service codes internally processed by the Cisco IP phone. You can delete any * code that you do not want the call server to process. Feature Dial Services Codes: Tells the Cisco IP phone what to do when the user is listening to the first or second dial tone.
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You can configure one or more * codes into this parameter, such as *72, or *72|*74|*67|*82, and so on. The maximum total length is 79 characters. When the user has a dial tone (first or second dial tone), they can enter a * code (and the following target number according to current dial plan) to trigger the Cisco IP phone to call the target number prepended by the * code. For example: a. After the user dials *72, the Cisco IP phone plays a special prompt tone while waiting for the user the enter a target number (which is validated according to the dial plan as in normal dialing). b. When a complete number is entered, the Cisco IP phone sends an INVITE to *72 target_number as in a normal call. This feature allows the proxy to process features such as call forward (*72) or BLock Caller ID (*67). 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 *72c|*67p. Following is a list of allowed dial tone parameters (note the use of back quotes surrounding the parameter without spaces). c = Cfwd dial tone d = Dial tone m = MWI dial tone o = Outside dial tone p = Prompt dial tone s = Second dial tone x = No tones are place, x is any digit not used above
If no tone parameter is specified, the Cisco IP phone plays the 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, add that * code in the dial plan.
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Activating or Deactivating Supplementary Services

You can disable services handled locally by the phone in one of two ways: Delete the star code in the Vertical Service Activation section in the Regional tab. Disable the service in the Phone tab. See Configuring Supplementary Services (Star Codes), page 181.
NOTE If a service is enabled in the Phone tab but cleared in the Regional tab, the service
can still be enabled/disabled by the end- user from the phone LCD or the configuration utility. If a service is disabled, the soft button associated with that service is hidden on the LCD. Also, any menu item associated with a disabled service is preceded with an exclamation mark (!). A supplementary service should be disabled if the user has not subscribed for it or the service provider intends to support similar service using other means than relying on the Cisco IP phone.
The Cisco SPA300 Series and Cisco SPA500 Series IP phones support all services that can be activated on a phone (call forward, do not disturb, and so on). Vertical service announcement codes apply only when the user dials the corresponding star code. Following is an example of how you can use these fields:
<Service Annc Base Number> = 1234 <Service Annc Extension Codes>= "CWT:00;CWF:01;FAT:02;FAF:05;FBT:03;FBF:05;FNT:04;FNF:05;" Here CWT: Call waiting service enabled; CWF: Call waiting service disabled; FAT: Call forward all service enabled; FAF: Call forward all service disabled; FBT: Call forward busy service enabled; FBF: Call forward busy service disabled; FNT: Call forward no answer enabled; FNF: Call forward no answer disabled;
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When the user enables call waiting service, the IP phone automatically calls "123400@$proxy". When the user disables the call waiting service, IP phone connects to "123401@$proxy". If the <Service Annc Extension Codes> do not define CWT/CWF extension codes, the IP phone defaults to normal.
Bonus Services Announcement description

When the user enables the callback service using the *code, the IP phone automatically calls 123400@$proxy. When the user disables the callback service using the *code, the IP phone automatically connects to the 123401@$proxy. If the Service Annc Extension Codes do not define CBT/CBF extension codes, the IP phone does not use this feature.
[Line1/2]<Service Announcement Serv> = Yes [Regional]<Service Annc Base Number> = {announcement server base number} [Regional]<Service Annc Extension Codes> = {SAEC Script} SAEC Script format:{SA_map;}* Here * means 0 or multiple SA_map syntax: SA_serv=SA_extcode SA_serv is the name of service plus the current condition; SA_extcode is the extension code which the ANNC server will route to. Appendix: SA_serv list 1) Call Back CBT: Call back enabled CBF: Call back disabled CBB: Call back busy enabled 2) Call Forward FAT: Call forward all enabled FAF: Call forward all disabled FBT: Call forward busy enabled FBF: Call forward busy disabled FNT: Call forward no answer enabled FNF: Call forward no answer disabled FLT: Call forward last enabled FLF: Call forward last disabled 3) Call Waiting CWT: Call waiting enabled CWF: Call waiting disabled 4) Block Last Call BLT: Block last call enabled BLF: Block last call disabled 5) Accept Last Call ALT: Accept last call enabled ALF: Accept last call disabled
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6) Block Caller ID BCT: Block caller id enabled BCF: Block caller id disabled 7) Distinctive Ringing DRT: Distinctive ringing enabled DRF: Distinctive ringing disabled 8) Speed Dial SDT: Speed dial enabled SDF: Speed dial disabled 9) Secure Call SCT: Secure call enabled SCF: Secure call disabled 10) Do Not Disturb DDT: DND enabled DDF: DND disabled 11) Caller ID CDT: Caller ID enabled CDF: Caller ID disabled 12) CW CID WDT: CWCID enabled WDF: CWCID disabled 13) Block Anonymous call BAT: Block anonymous call enabled BAF: Block anonymous call disabled
Outbound Call Codec Selection Codes

Codec call selection codes affect voice quality. For more information about voice codecs, see the Supported Codecs section on page 132. You can choose a preferred codec for a call or force a call to use a specific codec.
Prefer G.711u (*017110) through G.729a (*01729)Sets the preferred codec for next outbound call. If the preferred codec is unavailable, the second, then the third preferred codec is used, if specified (see the Configuring Voice Codecs section on page 136).
Force G.711u (*027110) through G.729a (*02729) Forces the specified codec for next outbound call. If the specified codec is unavailable, the preferred codecs are used in order, if specified (see the Configuring Voice Codecs section on page 136).
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This section contains both DTMF parameters and localization parameters: DTMF Parameters, page 189 Localizing Your IP Phone, page 190 Managing the Time and Date, page 191 Configuring Daylight Saving Time, page 192 Daylight Saving Time Examples, page 193 Selecting a Display Language, page 193 Creating a Dictionary Server Script, page 194
DTMF Parameters
Dual Tone Multi-Frequency (DTMF) is the system used by touch-tone phones. DTMF assigns a specific frequency (consisting of two separate tones) to each key so that it can easily be identified by a microprocessor. In-Band and Out-of-Band (RFC 2833): IP phones can relay DTMF digits as out-ofband events to preserve the fidelity of the digits. This can enhance the reliability of DTMF transmission required by many IVR applications such as dial-up banking and airline information. The following parameters can either help false detection or get better detection by the IVR. In general, the default values are recommended for both IVR functions.
DTMF Playback Level: Local DTMF playback level in decibels per minute, up
to one decimal place. Applicable locally when a user presses a digit or when the phone receives an out-of-band (OOB) DTMF signal from the network side. Does not affect DTMF transmission. Defaults to -16.
DTMF Playback Length: Local DTMF playback duration in milliseconds.

Affects only OOB. Defaults to .1.
Inband DTMF Boost: Controls the amount of amplification applied to DTMF

signals. Affects only tones sent by inband method. Choices are 0, 3, 6, 9, 12, 15, and 18 decibels. Defaults to 12 dB.
To help false detection, avoid inband and use OOB. With OOB, the DTMF Playback Length does not matter. If you use inband, use a smaller DTMF Boost value.
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To get better detection by the IVR, avoid inband and use OOB. This way, the DTMF tone is reconstructed by the PSTN gateway or the remote endpoint, and the quality is not subject to distortion from the audio codec. If you use OOB, use a slightly longer DTMF Playback Length. If you use inband, use a higher Inband DTMF boost.
NOTE On the Cisco SPA525G2, when using the Mobile Link line (through the Bluetooth-
enabled mobile phone), the local user can hear a double tone (echo) when pressing digits (DTMF tones) while engaged on a call. This can happen with certain mobile phones that have the option to play locally the local tone (which is also played by the Cisco SPA525G2). This does not affect operation with interactive voice response applications, as the tone is audible only on the local device. See the Cisco support community at http://www.cisco.com/go/smallbizsupport for phone compatibility information, and also consult the latest Cisco SPA525G2 release notes, available at cisco.com.

The following table describes the localization parameters in the Miscellaneous section.
Field
Set Local Date (mm/dd)
Description
Enter the local date (mm represents the month and dd represents the day). The year is optional and uses two or four digits. For example, May 1, 2008, can be entered as: 05/01 or 05/01/08 or 05/01/2008
Set Local Time (HH/mm) Time Zone
Enter the local time (hh represents hours and mm represents minutes). Seconds are optional. Selects the number of hours to add to GMT to generate the local time for caller ID generation. Choices are GMT-12:00, GMT11:00,, GMT, GMT+01:00, GMT+02:00, , GMT+13:00. Defaults to GMT-08:00.
Time Offset (HH/ mm)
This specifies the offset from GMT to use for the local system time.
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Field
Daylight Saving Time Rule Daylight Saving Enable Dictionary Server Script Language Selection
Description
Enter the rule for calculating daylight saving time. See the Configuring Daylight Saving Time section on page 192. Select yes to enable or no to disable DST on the phone. This setting affects all lines (extensions) on the phone. Defines the location of the dictionary server, the languages available and the associated dictionary. See the Creating a Dictionary Server Script section on page 194. Specifies the default language. The value must match one of the languages supported by the dictionary server. The script (dx value) is as follows: <Language_Selection ua="na"> </Language_Selection> Defaults to blank; the maximum number of characters is 512. For example: <Language_Selection ua="na"> Spanish </Language_Selection>
Managing the Time and Date

Cisco IP phones obtain the current time information in one of three ways: NTP ServerYou can configure one or two NTP servers on the phone. When the phone first boots up, it tries to contact the first NTP server to get the current time. The phone periodically synchronizes its time with the NTP server. The synchronization period is fixed at 1 hour. In between updates, the phone tracks time with its own internal clock. SIP MessagesEach SIP message (request or response) sent to the phone may contain a Date header with the current time information. If the header is present, the phone uses it to update its current time. Manual SetupThe phone also lets you manually enter the current time and date from the phone GUI or the configuration utility. However, this value is overridden by the NTP time or SIP Message Date whenever they are available to the phone. Manual setup requires you to enter the time in 24hour format only.
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The time served by the NTP Server and the SIP Date Header are expressed in GMT time. The local time is obtained by offsetting the GMT according to the time zone of the region. The Time Zone parameter can be configured from the web page or through provisioning. This time can be further offset by the Time Offset (HH/mm) parameter, which must be entered in 24-hour format. This parameter can also be configured from the phones LCD display.
NOTE The Time Zone and Time Offset (HH/mm) offset values are not applied to manual
time and date setup.
Configuring Daylight Saving Time

The phone supports auto adjustment for daylight saving time. You must set Daylight Savings Time Enable to yes and enter the DST rule. This option affects the time stamp on CallerID. To enter the rule for calculating DST, include the start, end, and save values separated by semi-colons (;) as follows:
Start = start-time; end=end-time; save = save-time
For example, the default DST rule is

start=4/1/7;end=10/-1/7;save=1.
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 month value equals any value in the range 1-12 (January-December). The day value equals any + or - 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. Optional time values: HH represents hours (0-23), mm represents minutes (0-59). and ss represents seconds (0-59). Optional values inside brackets [ ] are assumed to be 0 if not specified. Midnight is represented by 0:0:0 of the given date.
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The save-time value is the number of hours, minutes, and/or seconds to add to the current time during DST. The save-time value can be preceded by a plus (+) or minus (-) sign to indicate addition or subtraction.
Daylight Saving Time Examples

The following example configures daylight saving time for the U.S, adding one hour starting at midnight on the first Sunday in April and ending at midnight on the last Sunday of October; add 1 hour (USA, North America):
start=4/1/7/0:0:0;end=10/31/7/0:0:0;save=1 start=4/1/7;end=10/-1/7;save=1 start=4/1/7/0;end=10/-1/7/0;save=1
The following example configures daylight saving time for Egypt, starting at midnight on the last Sunday in April and ending at midnight on the last Sunday of September:
start=4/-1/7;end=9/-1/7;save=1 (Egypt)
The following example configures daylight saving time for New Zealand, starting at midnight on the first Sunday of October and ending at midnight on the third Sunday of March. This only applies to countries that recognize daylight saving time.
start=10/1/7;3/22/7;save=1 (New Zealand)
The following example reflects the new change starting March 2007. DST starts on the second Sunday in March and ends on the first Sunday in November:
start=3/8/7/02:0:0;end=11/1/7/02:0:0;save=1
Selecting a Display Language

This section describes how to localize the Cisco SPA300 and Cisco SPA500 Series IP Phone display language. You can define up to nine languages, in addition to English, to be available and host the dictionaries for each of the languages on the HTTP or TFTP provisioning server. Language support follows Cisco dictionary principles.
NOTE The WIP310 does not support localization; however, daylight saving time
adjustment is supported. Use the Language Selection parameter to select the phones default display language. The value must match one of the languages supported by the dictionary server. The script (dx value) is as follows:
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<Language_Selection ua="na"> </Language_Selection>
Defaults to blank; the maximum number of characters is 512. For example:

<Language_Selection ua="na"> Spanish </Language_Selection>
During startup, the phone checks the selected language and downloads the dictionary from the TFTP/HFTP provisioning server indicated in the phones configuration. The dictionaries are available at the support website. See Appendix C, Where to Go From Here, for the website location. The end user can change the language of the phone on the phone by following these steps:
STEP 1 Press the Setup button. STEP 2 Select Language, then press the Select soft button. STEP 3 Select Option to change the language. STEP 4 With the desired language selected, press Save.
Creating a Dictionary Server Script

The Dictionary Server Script defines the location of the dictionary server, the languages available and the associated dictionary. The syntax is as follows:
Dictionary_Server_Script ua="na"/Dictionary_Server_Script
Defaults to blank; the maximum number of characters is 512. The detailed format is as follows:
serv={server ip port and root path}; d0=language0;x0=dictionary0 filename; d1=language1;x1=dictionary1 filename; d2=language2;x2=dictionary2 filename; d3=language3;x3=dictionary3 filename; d4=language4;x4=dictionary4 filename; d5=language5;x5=dictionary5 filename; d6=language6;x6=dictionary6 filename; d7=language3;x7=dictionary7 filename; d8=language8;x8=dictionary8 filename; d9=language5;x9=dictionary9 filename;
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For example:
Dictionary_Server_Script ua="na" serv=tftp://192.168.1.119/ ;d0=English;x0=enS_v101.xml;d1=Spanish;x1=esS_v101.xml / Dictionary_Server_Script
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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. If the Cisco SPA IP Phones are part of the Cisco SPA 9000 Voice System, dial plans are configured on the Cisco SPA 9000. In installations where a Cisco SPA 9000 is not present (such as IP Centrex installations), installations where the phones are removed from the SPA 9000 (such as by a VPN), or other situations, dial plans can be configured on the IP phone using the configuration utility. For more information on using dial plans on the Cisco SPA 9000 Voice System, see the Cisco SPA 9000 Voice System Administration Guide. See the Appendix C, Where to Go From Here, for the location of the document. 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, page 196 Editing Dial Plans on the IP Phone, page 206
About Dial Plans

This section provides information to help you understand how dial plans are implemented. The Cisco SPA IP Phones and the SPA 9000 are involved in applying various levels of the dial plans and process the digits sequence in the same manner.
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When a user lifts a handset or presses a speaker button on the IP phone, the following sequence of events begins: 1. The phone begins collecting the dialed digits. The inter-digit timers starts tracking the time that elapses between digits. 2. If the inter-digit timer value is reached, or if another terminating event occurs, the phone compares the dialed digits with the IP phones dial plan. (This dial plan is configured in the configuration utility in the Voice tab, on the tab for each extension (Ext N), under the Dial Plan section.) If the phone is part of a Cisco SPA 9000 system: 3. If the phone dial plan allows the call to process, the dialed numbers are sent to the Cisco SPA 9000. 4. The Cisco SPA 9000 compares the dialed digits to the CALL ROUTING RULE (on the SPA 9000 Voice > SIP page, PBX Parameters section). 5. If the call routing rule allows the call to process, then the Cisco SPA 9000 compares the dialed digits to the LINE INTERFACE dial plan (on the Cisco SPA 9000 Voice > Line N page, Dial Plan section). 6. The Cisco SPA 9000 uses the information in the line dial plan to manipulate the number (for example, to remove steering digits) and then transmits the number.
NOTE The dial plan feature (digit sequences and timers) is not used with the Cisco
SPA525G2 phone line associated to Mobile Link (a Bluetooth-enabled mobile phone). Mobile phone dial plan rules continue to apply in this scenario. Refer to the following topics: Digit Sequences, page 198 Digit Sequence Examples, page 200 Acceptance and Transmission of the Dialed Digits, page 202 Dial Plan Timer (Off-Hook Timer), page 203 Interdigit Long Timer (Incomplete Entry Timer), page 204 Interdigit Short Timer (Complete Entry Timer), page 205
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Digit Sequences
A dial plan contains a series of digit sequences, separated by the | character. The entire collection of sequences is enclosed within parentheses. Each digit sequence within the dial plan consists of a series of elements, which are individually matched to the keys that the user presses.
NOTE White space is ignored, but may be used for readability.
Digit Sequence 0 1 2 3 4 5 6 7 8 9 0 * # x [sequence]
Function
Enter any of these characters to represent a key that the user must press on the phone keypad. Enter x to represent any character on the phone keypad. Enter characters within square brackets to create a list of accepted key presses. The user can press any one of the keys in the list.
Numeric range For example, you would enter [2-9] to allow the user to press any one digit from 2 through 9. Numeric range with other characters For example, you would enter [35-8*] to allow the user to press 3, 5, 6, 7, 8, or *.
.
(period)
Enter a period for element repetition. The dial plan accepts 0 or more entries of the digit. For example, 01. allows users to enter 0, 01, 011, 0111, and so on.
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Digit Sequence <dialed:substituted> Function
For sequence substitution, use this format to indicate that certain dialed digits are replaced by other characters when the sequence is transmitted. The dialed digits can be zero or more characters. EXAMPLE 1: <8:1650>xxxxxxx When the user presses 8 followed by a seven-digit number, the system automatically replaces the dialed 8 with 1650. If the user dials 85550112, the system transmits 16505550112. EXAMPLE 2: <:1>xxxxxxxxxx In this example, no digits are replaced. When the user enters a 10-digit string of numbers, the number 1 is added at the beginning of the sequence. If the user dials 9725550112, the system transmits 19725550112
,
(comma)
For an intersequence tone, enter a comma between digits to play an outside line dial tone after a userentered sequence. EXAMPLE: 9, 1xxxxxxxxxx An outside line dial tone is sounded after the user presses 9, and the tone continues until the user presses 1.
!
(exclamation point)
For number barring, enter an exclamation point to prohibit a dial sequence pattern. EXAMPLE: 1900xxxxxxx! The system rejects any 11-digit sequence that begins with 1900.
*xx S0 or L0
Enter an asterisk to allow the user to enter a 2-digit star code. For Interdigit Timer Master Override, 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. For a pause, enter P followed by a number and a space. The duration of the pause is the specified number of seconds. This feature is typically used for implementation of a hot line and warm line, with 0 delay for the hot line and a non-zero delay for a warm line. EXAMPLE: P5
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sequences entered in the regional parameter settings to the end of the dial plan. Likewise, if Enable_IP_Dialing is enabled, then IP dialing is also accepted on the associated line.
NOTE The Cisco SPA 9000 and the Cisco IP phones implicitly append the vertical code
Digit Sequence Examples

The following examples show digit sequences that you can enter in a dial plan. In a complete dial plan entry, sequences are separated by a pipe character (|), and the entire set of sequences is enclosed within parentheses. EXAMPLE: ( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9, 011xxxxxx. | 0 | [49]11 ) Extensions on your system EXAMPLE: ( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9, 011xxxxxx. | 0 | [49]11 )
[1-8]xx Allows a user dial any three-digit number that starts with the digits 1 through 8. If your system uses four-digit extensions, you would instead enter the following string: [1-8]xxx
Local dialing with seven-digit number EXAMPLE: ( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9, 011xxxxxx. | 0 | [49]111)
9, xxxxxxx After a user presses 9, an external dial tone sounds. The user can
enter any seven-digit number, as in a local call. Local dialing with 3-digit area code and a 7-digit local number EXAMPLE: ( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9, 011xxxxxx. | 0 | [49]11 )
9, <:1>[2-9]xxxxxxxxx This example is useful where a local area code is
required. After a user presses 9, an external dial tone sounds. The user must enter a 10-digit number that begins with a digit 2 through 9. The system automatically inserts the 1 prefix before transmitting the number to the carrier.
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Local dialing with an automatically inserted 3-digit area code EXAMPLE: ( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9, 011xxxxxx. | 0 | [49]11 )
8, <:1212>xxxxxxx This 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 EXAMPLE: ( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9, 011xxxxxx. | 0 | [49]11 )
9, 1 [2-9] xxxxxxxxx After the user presses 9, an external dial tone sounds. The
user can enter any 11-digit number that starts with 1 and is followed by a digit 2 through 9. Blocked number EXAMPLE: ( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9, 011xxxxxx. | 0 | [49]11 )
9, 1 900 xxxxxxx ! This digit sequence is useful if you want to prevent users
from dialing numbers that are associated with high tolls or inappropriate content, such as 1-900 numbers in the U.S.. After the user 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 EXAMPLE: ( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9, 011xxxxxx. | 0 | [49]11 )
9, 011xxxxxx. After the user presses 9, an external dial tone sounds. The user
can enter any number that starts with 011, as in an international call from the U.S. Informational numbers
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EXAMPLE: ( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9, 011xxxxxx. | 0 | [49]11 )
0 | [49]11 This example includes two digit sequences, separated by the pipe
character. The first sequence allows a user to dial 0 for an operator. The second sequence allows the user to enter 411 for local information or 911 for emergency services.
Acceptance and Transmission of 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 IP PBX either accepts the user-dialed sequence and initiates a call, or else rejects the sequence as invalid. The user hears the reorder (fast busy) tone if the dialed sequence is invalid. The following table explains how terminating events are processed. Terminating Event
The dialed digits do not match any sequence in the dial plan. The dialed digits exactly match one sequence in the dial plan.
Processing
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.
A timeout occurs.
The number is rejected if the dialed digits are not matched to a digit sequence in the dial plan within the time specified by the applicable interdigit timer.
The Interdigit Long Timer applies when the dialed digits do not match any digit sequence in the dial plan. The default value is 10 seconds. The Interdigit Short Timer applies when the dialed digits match one or more candidate sequences in the dial plan. The default value is 3 seconds.
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Terminating Event
The user presses the # key or the dial softkey on the phone display.
Processing
If the sequence is complete and is allowed by the dial plan, the number is accepted and is transmitted according to the dial plan. If the sequence is incomplete or is blocked by the dial plan, the number is rejected.
Dial Plan Timer (Off-Hook Timer)

You can think of the Dial Plan Timer as the off-hook timer. This timer starts counting when the phone goes off hook. If no digits are dialed within the specified number of seconds, the timer expires and the null entry is evaluated. Unless you have a special dial plan string to allow a null entry, the call is rejected. The default value is 5.
Syntax for the Dial Plan Timer

SYNTAX: (P s<:n> | dial plan ) s: The number of seconds; if no number is entered after P, the default timer of 5 seconds applies. n: (optional): The number to transmit automatically when the timer expires; you can enter an extension number or a DID number. No wildcard characters are allowed because the number will be transmitted as shown. If you omit the number substitution, <:n>, 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: (P9 | (9,8<:1408>[2-9]xxxxxx | 9,8,1[29]xxxxxxxxx | 9,8,011xx. | 9,8,xx.|[1-8]xx)
P9 After taking a phone off hook, a user has 9 seconds to begin dialing. If no
digits are pressed within 9 seconds, the user hears a reorder (fast busy) tone. By setting a longer timer, you allow more time for users to enter the digits. Create a hotline for all sequences on the System Dial Plan EXAMPLE: (P9<:23> | (9,8<:1408>[2-9]xxxxxx | 9,8,1[29]xxxxxxxxx | 9,8,011xx. | 9,8,xx.|[1-8]xx)
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P9<:23> After taking the phone off hook, a user has 9 seconds to begin
dialing. If no digits are pressed within 9 seconds, the call is transmitted automatically to extension 23. Create a hotline on a line button for an extension EXAMPLE: ( P0 <:1000>) With the timer set to 0 seconds, the call is transmitted automatically to the specified extension when the phone goes off hook. Enter this sequence in the Phone Dial Plan for Ext 2 or higher on a client station.
Interdigit Long Timer (Incomplete Entry Timer)

You can think of this timer as the incomplete entry timer. This timer measures the interval between dialed digits. It applies as long as the dialed digits do not match any digit sequences in the dial plan. Unless the user enters another digit within the specified number of seconds, the entry is evaluated as incomplete, and the call is rejected. The default value is 10 seconds.
NOTE This section explains how to edit a timer as part of a dial plan. Alternatively, you can
modify the Control Timer that controls the default interdigit timers for all calls. See Resetting the Control Timers, on page 207.
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.
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Interdigit Short Timer (Complete Entry Timer)
You can think of this timer as the complete entry timer. This timer measures the interval between dialed digits. It applies when the dialed digits match at least one digit sequence in the dial plan. Unless the user enters another digit within the specified number of seconds, the entry is evaluated. If it is valid, the call proceeds. If it is invalid, the call is rejected. The default value is 3 seconds.
Syntax for the Interdigit Short Timer

SYNTAX 1: S :s, ( dial plan ) Use this syntax to apply the new setting to the entire dial plan within the parentheses. SYNTAX 2: sequence Ss Use this syntax to apply the new setting to a particular dialing sequence. s: The number of seconds; if no number is entered after S, the default timer of 5 seconds applies.
Examples for the Interdigit Short Timer

Set the timer for the entire dial plan. EXAMPLE: S:6, (9,8<:1408>[2-9]xxxxxx | 9,8,1[29]xxxxxxxxx | 9,8,011xx. | 9,8,xx.|[1-8]xx)
S:6, While entering a number with the phone off hook, a user can pause for
up to 15 seconds between digits before the Interdigit Short Timer expires. This setting is especially helpful to users such as sales people, who are reading the numbers from business cards and other printed materials while dialing. Set an instant timer for a particular sequence within the dial plan. EXAMPLE: (9,8<:1408>[2-9]xxxxxx | 9,8,1[2-9]xxxxxxxxxS0 | 9,8,011xx. | 9,8,xx.|[1-8]xx)
9,8,1[2-9]xxxxxxxxxS0 With the timer set to 0, the call is transmitted
automatically when the user dials the final digit in the sequence.
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Editing Dial Plans on the IP Phone

You can edit dial plans and can modify the control timers. To edit the dial plans on the IP phone:
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click the Ext N tab, where N is the extension being configured. STEP 4 In the Dial Plan section, enter the digit sequences in the Dial Plan field. For more
information and examples, see Digit Sequences, on page 198. The default (US-based) system-wide dial plan appears automatically in the field. You can delete digit sequences, add digit sequences, or replace the entire dial plan with a new dial plan. For more information and examples, see Digit Sequences, on page 198. Separate each digit sequence with a pipe character, and enclose the entire set of digit sequences within parentheses. Refer to the following example: (9,8<:1408>[2-9]xxxxxx | 9,8,1[2-9]xxxxxxxxx | 9,8,011xx. | 9,8,xx.|[1-8]xx)
STEP 5 (Optional) Enter the Caller ID MapInbound caller ID numbers can be mapped to a
different string. For example, a number that begins with +44xxxxxx can be mapped to 0xxxxxx. This feature has the same syntax as the Dial Plan parameter. With this parameter, you can specify how to map a caller ID number for display on screen and recorded into call logs. (Not applicable to WIP310.)
STEP 6 (Optional) Enable IP dialingEnable or disable IP dialing. Defaults to no. STEP 7 (Optional) Emergency NumberEnter a comma-separated list of emergency
numbers. When one of these numbers is dialed, the unit disables processing of CONF, HOLD, and other similar softkeys or buttons to avoid accidentally putting the current call on hold. The phone also disables hook flash event handling. Only the far end can terminate an emergency call. The phone is restored to normal after the call is terminated and the phone is back on-hook. Maximum number length is 63 characters. Defaults to blank (no emergency number). (Not applicable to WIP310.)
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Resetting the Control Timers
STEP 9 If you need to configure a dial plan for any other extensions on the phone
(depending on the model), click the appropriate Extension tab, enter the dial plan, and submit the changes.
STEP 10 Verify that you can successfully complete a call using each digit sequence that
you entered in the dial plan.

NOTE If you hear a reorder (fast busy) tone, you need to review your entries and modify
the dial plan appropriately. See Digit Sequences, on page 198.
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 196.
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login and advanced. STEP 3 Click Voice tab > Regional. STEP 4 Scroll down to the Control Timer Values section. STEP 5 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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9
Configuring the Cisco SPA500S Attendant Console
The Cisco SPA500S is a 32-button attendant console for the Cisco SPA500 Series IP Phones. The Cisco SPA500S works in both SIP and SPCP mode. The Cisco SPA500S connects to the phone with the attachment arm provided (not shown). It obtains power directly from the phone and does not require a separate power supply. Two Cisco SPA500S units can be attached to a single phone to monitor a total of 64 separate lines. For more information about installing and configuring the Cisco SPA500S with a Cisco SPA 9000 system, see the Cisco SPA 9000 Voice System Installation and Configuration Guide. This chapter contains the following sections: Configuring the Cisco SPA 9000 for the Cisco SPA500S, page 211 Configuring the BroadSoft Server for the Cisco SPA500S, page 211 Configuring the Asterisk Server for the Cisco SPA500S, page 212 Configuring the Cisco SPA500S, page 213 Unit/Key Configuration Scripts, page 214 Attendant Console Parameters, page 217 Monitoring the Cisco SPA500S, page 219
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Cisco SPA500S Features
Cisco SPA500S Features

Cisco SPA500S features include: 32 programmable speed-dial or direct station select (DSS) buttons/LEDs Three-color LEDs (red, green, and orange) Support for BroadSoft Busy Lamp Field (BLF) Support for Line Monitoring Illuminated line status monitoring (Idle, Ringing, Busy, Null, or Registration Error) One-touch Call TransferIncoming calls can be immediately transferred to the target destination by pressing a button/key as assigned on the Cisco SPA500S.
The following table describes Cisco SPA500S ports and LEDs.

. Port/LED AUX IN AUX OUT Solid Green Solid Red Blinking Red Orange Off Meaning Connects to the phone. Connects to a second Cisco SPA500S unit (optional) Idle In-use Ringing Solid: not registered. Blinking: configuration error. Not configured
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Setting Up the Cisco SPA500S Attendant Console

To configure the server to use the Cisco SPA500S, configure each extension that will be monitored by the attendant console. The Cisco SPA500S is a SIP subscriber in relation to each SIP proxy server, which allows the Cisco SPA500S to receive NOTIFY messages from the SIP proxy that indicate the status of each monitored phone. In general, the SIP proxy is identified by its IP address, or through a hostname if DNS is configured. The way in which each phone and extension is identified is vendor-specific. To set up the Cisco SPA500S attendant console:
STEP 1 Complete the physical installation of the Cisco SPA500S unit, which connects it to
the phone. For instructions on installing the Cisco SPA500S and an introduction to its use, refer to the Cisco SPA500S Quick Start Guide or the phone user guides available on Cisco.com. See Appendix C, Where to Go From Here, for a list of those guides.
STEP 2 Configure one of the following SIP proxy servers:
Cisco SPA 9000See Configuring the Cisco SPA 9000 for the Cisco SPA500S, page 211 (verify that your version of the Cisco SPA 9000 supports the Cisco SPA500S). BroadSoftSee Configuring the BroadSoft Server for the Cisco SPA500S, page 211. AsteriskSee Configuring the Asterisk Server for the Cisco SPA500S, page 212.
STEP 3 Configure the Cisco SPA500S using the configuration utility. The phone to which
the Cisco SPA500S is physically attached and the computer running the configuration utility must be on the same network.
NOTE CTI must be enabled on the phone for an attached Cisco SPA500S to
properly monitor the IP phones line status when the SIP proxy server type is set to Cisco SPA 9000. See the Configuring SIP section on page 89.
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Configuring the Cisco SPA 9000 for the Cisco SPA500S

You must still enable Computer Telephony Integration (CTI) on the Cisco SPA 9000 configuration utility, as shown in the following procedure:
STEP 1 Connect to the web-based configuration utility for the Cisco SPA 9000. STEP 2 Click Admin Login on the configuration utility page. STEP 3 If necessary, enter admin and your password, then Advanced. STEP 4 Click the SIP tab. STEP 5 In the SIP Parameters section, select yes from the CTI Enable drop-down list. STEP 6 Click Submit All Changes.
Configuring the BroadSoft Server for the Cisco SPA500S

The BroadSoft server requires you to assign IP phone users to the Busy Lamp (BLF) Monitor Users List. The BroadSoft server sends updates on the status of each phone on this list to the Cisco SPA500S, which subscribes for this service. To configure the BroadSoft server to support the Cisco SPA500S:
STEP 1 On the BroadSoft server, in the user profile used by the phone to register with the
BroadSoft server, select Assign Services.

STEP 2 On the Assign Services page, from the Available Services column, move Busy
Lamp to the User Services column.

STEP 3 Define the List URI used by the Cisco SPA500S to register for BLF monitoring
service with the BroadSoft server. This value must match the value entered following the sub = keyword (for example, cisco_list). Select the domain from the drop-down list to match the Unit Key.
NOTE If you configure more than one monitored list on the BroadSoft server, use the vid=
keyword in each unit/keyconfiguration script to specify the phone extension to use for each list.
STEP 4 On the Busy Lamp Field page, move users that need to be monitored from the
Available Users column to the Monitored Users column.

STEP 5 Click the Add (or Add All) button to move each user to the Monitored Users
column. The Directory Number (DN) associated with each user account when it is
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created on the BroadSoft Server is shown in parenthesis in the Monitored Users list. You use this DN to identify the specific phone assigned to each key on the Cisco SPA500S.
STEP 6 Save and enable your configuration changes on the BroadSoft server.
See also the Configuring BroadSoft Busy Lamp Field Auto-Configuration section on page 217.
Configuring the Asterisk Server for the Cisco SPA500S

To configure the Asterisk server to allow the Cisco SPA500S to register for BLF monitoring:
STEP 1 Add a context in the extensions.conf file. STEP 2 Add a Subscribecontext command to point to the context in the sip.conf file. STEP 3 Configure the Cisco SPA500S to register with the Asterisk server (see
Configuring the Cisco SPA500S section on page 213. The following example context uses home for extension 3500. This is entered in the file extensions.conf:
[home] exten => 3500,1,Dial(SIP/3500) exten => 3500,hint,SIP/3500 exten => 3500,2,Voicemail,u3500 exten =>3500,1,3,hangup ...
In the following example, extension 3500 is used to add Subscribecontext to point to the context.
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This is entered in the file sip.conf:

[3500] type=friend secret=3500 callerid="spa3500"<3500> nat=no context=home mailbox=3500 Subscribecontext=home ... ...
Configuring the Cisco SPA500S

Complete the configuration required for the extensions on your IP PBX and obtain the following information: IP PBX IP address or other hostname Phone extension numbers
To configure the Cisco SPA500S:

NOTE Steps 1 through 4 are only required when the Server Type field is set to SPA9000.
These steps are not required for other server types such as BroadSoft and Asterisk.
STEP 1 Connect to the configuration utility for the phone to which the Cisco SPA500S is
connected.
STEP 2 Click Admin/Advanced on the configuration utility page. STEP 3 Click the SIP tab. STEP 4 Select yes from the CTI Enable drop-down list. STEP 5 Click the Attendant Console tab. STEP 6 Select yes from the Unit 1 Enable drop-down list. If you have installed two Cisco
SPA500S units, also select yes from the Unit 2 Enable drop-down list.
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STEP 7 Select your Server Type from the drop-down list:
SPA9000 BroadSoft Asterisk
STEP 8 Make sure that no is selected for Test Mode Enable. This option is disabled by
default. You cannot complete the software configuration for the Cisco SPA500S if this option is enabled. (You can use Test Mode Enable later to test the Cisco SPA500S.)
STEP 9 Create a configuration script for each target extension or user you want to monitor
using the Cisco SPA500S. Enter this script in the appropriate field for each unit/ key. See Unit/Key Configuration Scripts section on page 214.
Unit/Key Configuration Scripts

By default, all LEDs on the Cisco SPA500S are assigned to the first configured extension on the phone to which it is connected. Assign the LEDs to any of the other five extensions on the phone that you want to monitor using the Cisco SPA500S. The configuration script is composed of the following keywords, followed by an equal sign (=) and separated by semicolons (;): fncdefines which of the following functions are enabled for the specified key [separate more than one function with a plus sign (+)]: blfbusy lamp field function used for monitoring line activity sdspeed dial function cpcall pickup (if supported by the SIP proxy server). Call pickup (cp) must be supported by the SIP proxy server and be used with blf in the configuration. The syntax is fnc=blf+cp.
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subUse this keyword to identify the phones to be monitored). Its value and syntax is stationName@$PROXY, where system variable $PROXY contains the proxy server IP and port (e.g. 192.168.8.101:6060).
NOTE Unit/key LEDs will not light without the sub keyword. (Not required
for speed dials.) usr or ext (optional)Use one of these keywords to identify the specific users or extensions to be monitored. Its value and syntax is extensionNumber@$PROXY, where system variable $PROXY contains the proxy server IP and port (e.g. 192.168.8.101:6060). (The usr and ext keywords are interchangeable.) If the ext parameter is not used, all extensions on the phone are monitored. nme (optional) Use this field with the Cisco SPA 9000 to identify any alias that has been assigned to the extension in the IP phone configuration.The nme parameter indicates the extension name, which in this case is the same as the station name. vid (optional) All LEDs on the Cisco SPA500S use phone extensions that they are assigned to. By default, LEDs on the Cisco SPA500S are assigned to the first configured extension on the connected phone. You can optionally assign LEDs to any other phone extensions using vid=keyword. Use this field to identify the phone extension to use with the monitored list specified by the sub= keyword, when more than one BLF monitored list is configured on the SIP proxy server. The possible values are 1 to 6, corresponding to each of the six extensions available on the phone. Only use the vid= keyword in the first entry assigned to each phone extension. Subsequent keys will use the same extension. See Attendant Console Parameters section on page 217.
Assigning Cisco SPA500S LEDs to Phone Extensions

By default, all 32 keys on the Cisco SPA500S are assigned to the first extension on the IP phone (extension 101 based on the default multicast autoconfiguration). To limit the number of LEDs assigned to the first extension, use the vid=1 keyword. Then use the vid=2 keyword to assign the next set of keys to the second extension. The numeric values correspond to the default extensions on the phone as follows: vid=1extension 101 vid=2extension 102 vid=3extension 103 vid=4extension 104
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vid=5extension 105 vid=6extension 106
Cisco SPA 9000 Syntax

Find more complete information in the Cisco SPA 9000 Voice System Installation and Configuration Guide. The following entry enables speed dialing, BLF monitoring, and call pick up on a Cisco SPA 9000 server with: IP address 192.168.1.101 station name phone1 extension 101 Cisco SPA500S unit/key using the phones extension 2
Example:
fnc=sd+blf+cp;sub=phone1@$PROXY;usr=101@$PROXY;nme=phone1;vid=2
BroadSoft syntax
For example, the following enables speed dialing and BLF monitoring, with a BLF monitoring list URI of marketing, for the user account reception, on a BroadSoft server with the IP address 192.168.100.1:
fnc=sd+blf;[email protected];[email protected]
The nme keyword is not used because the BroadSoft server uses the user account name assigned to the BLF monitoring list. Note that you can configure a list of BLF subscriptions automatically using a URI (rather than individually configuring each BLF entry). See the Configuring BroadSoft Busy Lamp Field Auto-Configuration section on page 217.
Asterisk syntax
The following is an example entry for a Asterisk server. This entry enables speed dialing, BLF monitoring, and call pickup on a Asterisk server with the IP address 192.168.1.11:
fnc=sd+blf+cp;[email protected];nme=35890
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Configuring BroadSoft Busy Lamp Field Auto-Configuration

Rather than configuring each BLF key individually, you can enter a single SUBSCRIBE URI and automatically generate configuration scripts for BLF keys for all users on the monitored list. To configure BLF auto-configuration:
STEP 1 Log in to the configuration utility. STEP 2 Click Admin Login. STEP 3 Click advanced. STEP 4 Click the Att Console tab. STEP 5 In the BLF List URI field, enter the URI to generate BLF keys:
[email protected];vid=2
Where: listname: Name of the list. domain.com: Name of the domain. vid: Assigns keys to the extension. By default, the SUBSCRIBE is sent when Ext 1 is registered. You must set the vid= to a different value to change this behavior.
For example:
[email protected];vid=2
Attendant Console Parameters

Parameters in the Attendant Console web page are described in the following table. For information about Unit/Key syntax, see the previous sections.
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9
Description
Specifies how long the subscription remains valid. After the specified period of time, elapses, the Cisco SPA500S initiates a new subscription. Defaults to 1800.
Parameter
Subscribe Expires
Subscribe Retry Interval Unit 1 Enable
Specifies the length of time to wait to try again if subscription fails. Enables or disables the first Cisco SPA500S unit (each phone can have up to two Cisco SPA500S units attached). Enables or disables the second Cisco SPA500S unit (each phone can have up to two Cisco SPA500S units attached). Length of delay before attempting to subscribe. Defaults to 1.
Unit 2 Enable
Subscribe Delay
Server Type Test Mode Enable
Selects the type of server used (SPA9000, BroadSoft, or Asterisk). Enables or disables test mode. When test mode is enabled, the LEDs are turned on when keys are pressed, going from off to green to red, and back to off. In test mode, when all the buttons on the attendant console are returned to off, all the keys become orange. The phone must be rebooted after the test is completed. The star code used for picking up a ringing call. Defaults to *98 (group call pickup). For directed call pickup, enter *97#. The user enters *97 and the extension to pick up.
Attendant Console Call Pickup Code
BLF List URI Unit 1 Key 1-32
Automatically configures BLF subscriptions for all users on a monitored list. Enter a strings that define the extension and other parameters associated with each lighted button on the first Cisco SPA500S unit. Keywords and values are case-sensitive. The configuration script is described in Unit/Key Configuration Scripts section on page 214.
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To display the status of the Cisco SPA500S, click the Attendant Console Status link on the GUI. The status of each attendant console attached (Unit 1 and Unit 2) is shown. Parameters are read-only.
Cisco SPA500S Unit Monitoring Notes

The following table describes each parameter; both units display the same parameters.
Parameter Unit Enable Subscribe Expires
Description Displays if the Unit is enabled or disabled. Displays when the current subscription expires. After the subscription expires, the Cisco SPA500S automatically requests a new subscription. Displays the version of the hardware. Displays whether the unit is connected or not. Displays the length of time the Cisco SPA500S waits to try again if subscription fails. Displays the version of the software currently running on the unit. Displays the name assigned to each key (1-32) on the Cisco SPA500S attendant console unit. Displays the function enabled for each key (1-32) on the Cisco SPA500S attendant console unit. Displays the extension assigned to each key (1-32) on the Cisco SPA500S attendant console unit. Displays the subscribe URI configured for each key (132) on the Cisco SPA500S attendant console unit. Displays the subscription status of the unit/key. The value can be Yes, Fail, or No. No indicates that the feature/function (fnc) of that line does not require a subscription (such as speed dial).
HW Version Unit Online Subscribe Retry Interval SW Version Key Name Type Line Station Subscribe
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A
Creating an LED Script
LED Script
The LED script describes the color and blinking pattern of a Line Key LED. Each script contains a number of fields separated by a semicolon(;). White spaces are ignored. Each field has the syntax <field-name> = <field-value>. The allowed field-name and corresponding field-values are listed below:
c=o|r|g|a
This field sets the color of the LED. The 4 choices are: o = off r = red g = green a = amber (orange)
p=n[b]|s[b]|f[b]|d[b]|u[d]
This field sets the blinking pattern of the LED. The 4 choices are: nb = no blink (steady on or off) sb = slow blink (1s on and 1s off) fb = fast blink (100ms on and 100ms off) ud = user-defined (according to the contents of the u field)
u=on/off/on/off/etc.
This is a user-defined blinking pattern used only when p = ud. It consists of up to 4 pairs of on/off duration in seconds with up to 2 decimal places; each value is separated by a forward slash (/).
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A
LED Script Examples
Example 1
c=r;p=sb
Color is red and pattern is slow blink. Example 2

c=o
LED is off. Example 3

c=g
Color is green and pattern is steady on (default). Example 4

c=a;p=ud;u=.1/.1/.1/.1/.1/.9
Color is amber (orange) and the blink pattern is: 100ms on, 100ms off, 100ms on, 100ms off, 100ms on, 900ms off
LED Pattern
The administrator can also specify a different color and pattern for each of the following states of the call appearance. Idle: This call appearance is not in use. It can be used to make outbound call on this station Local Seized: This call appearance has been seized by this station to prepare for an outbound call Local Progressing: This station is making an outbound call that is progressing Local Active: This station is engaged in a connected call on this call appearance Local Ringing: This station is ringing for an incoming call on this call appearance Local Held: This station has placed this call appearance on hold
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A
Remote Seized: This call appearance has been seized by another station to prepare for an outbound call Remote Progressing: Another station is making a call on this call appearance and is progressing Remote Active: Another station is engaged in a connected call on this call appearance Remote Ringing: Another station is ringing for an incoming call to this call appearance Remote Held: Another station has placed this call appearance on hold Remote Undefined: The share call state is not known (this station is waiting for a notification from the application server) Registration Failed: This station has failed to register with the proxy server for the corresponding extension Registering: The station is attempting registration with the proxy server for the corresponding extension. Disabled: This line key on this station is disabled Call Back: A call back (repeat dialing) operation is currently active on this call appearance
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B
Cisco SPA IP Phone Field Reference
This appendix describes the fields within the following sections of the configuration utility: Voice Tab Info Tab, page 224 System Tab, page 233 SIP Tab, page 240 Provisioning Tab, page 254 Regional Tab, page 255 Phone Tab, page 274 Ext Tab, page 294 User Tab, page 311 Attendant Console Status, page 317 Cisco SPA525G/525G2-Specific Tabs, page 318
NOTE For information about the Provisioning page, see the Cisco Small Business IP
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Info Tab
Info Tab
This section describes the fields for the following headings on the Info tab: System Information, page 224 Network Configuration (SPCP), page 226 VPN Status (SPA525G/525G2 Only), page 227 Product Information, page 227 Phone Status, page 228 Line/Call Status, page 230
NOTE The fields on this tab are read-only and cannot be edited.
System Information
Parameter Connection Type
Description Indicates the type of internet connection for the phone:

Current IP
DHCP Static IP PPPoE (only applicable to Cisco SPA525/525G2)
Displays the current IP address assigned to the IP phone. Displays the current host name assigned to the IP phone (defaults to SipuraSPA). Displays the network domain name of the IP phone. Displays the network mask assigned to the IP phone. Displays the default router assigned to the IP phone. Displays the primary DNS server assigned to the IP phone. Displays the secondary DNS server assigned to the IP phone.
Host Name
Domain Current Netmask Current Gateway Primary DNS
Secondary DNS
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Info Tab
B
Description Shows if Network Time Protocol is enabled.
Parameter NTP Enable (Cisco SPA525G/ 525G2 only) Primary NTP Server (Cisco SPA525G/525G2 only) Secondary NTP Server (Cisco SPA525G/525G2 only) TFTP Server (Cisco SPA525G/525G2 only) Bluetooth Enabled (Cisco SPA525G/525G2 only) Bluetooth Firmware Version (Cisco SPA525G/525G2 only) Bluetooth Connected (Cisco SPA525G/525G2 only) Bluetooth MAC (Cisco SPA525G/525G2 only) Connected Device ID (Cisco SPA525G/525G2 only) Wireless Enabled (Cisco SPA525G/525G2 only) Wireless Connected (Cisco SPA525G/525G2 only) Wireless MAC (Cisco SPA525G/525G2 only) SSID (Cisco SPA525G/ 525G2 only) Standard Channel (Cisco SPA525G/525G2 only) Security Mode (Cisco SPA525G/525G2 only)
IP Address of the primary NTP server.
IP Address of the secondary NTP server.
Address of the TFTP server for provisioning.
Shows if Bluetooth is enabled.
Displays the Bluetooth firmware version.
Shows if a Bluetooth device is connected to the phone.
Shows the hardware address of the Bluetooth device.
Shows the name of the connected Bluetooth device.
Shows if Wireless-G is enabled on the phone.
Shows if the phone is connected to the wireless network. Shows the hardware address of the Wireless-G controller. Shows the SSID, or name of the wireless router to which the phone is connected. Shows the wireless channel being used in the wireless connection. Shows if wireless security is configured on the phone (yes or no).
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Info Tab
B
Description Shows if Network Time Protocol is enabled. IP Address of the primary NTP server. IP Address of the secondary NTP server. Address of the TFTP server for provisioning. Shows if Bluetooth is enabled. Displays the Bluetooth firmware version. Shows if a Bluetooth device is connected to the phone. Shows the hardware address of the Bluetooth device. Shows the name of the connected Bluetooth device. Shows if Wireless-G is enabled on the phone. Shows if the phone is connected to the wireless network. Shows the hardware address of the Wireless-G controller. Shows the SSID, or name of the wireless router to which the phone is connected. Shows the wireless channel being used in the wireless connection. Shows if wireless security is configured on the phone (yes or no).
Cisco SPA525G/525G2-Specific Parameters:
Parameter NTP Enable Primary NTP Server Secondary NTP Server TFTP Server Bluetooth Enabled Bluetooth Firmware Version Bluetooth Connected Bluetooth MAC Connected Device ID Wireless Enabled Wireless Connected
Wireless MAC
SSID
Standard Channel
Security Mode
Network Configuration (SPCP)
Parameter TFTP Server
Description Address of the TFTP server for provisioning.
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Info Tab
B
Description IP address of the Unified Communications server. Populated by the Unified Communications Server; points to the directory application server. Populated by the Unified Communications Server; points to the Cisco XML application server. Populated by the Unified Communications Server; points to the authentication server. Populated by the Unified Communications Server; indicates if the DHCP address has been released.
Parameter Call Manager Directories URL
Services URL
Authentication URL
DHCP Address Released
VPN Status (SPA525G/525G2 Only)
Parameter VPN Connected Client Address Client Netmask Bytes Sent Bytes Recv
Description Indicates if the phone is connected to a VPN. IP address given to the phone from the VPN server. Netmask given to the phone from the VPN server. Size of data sent from the phone. Size of data received by the phone.
Product Information
Parameter Product Name Serial Number Software Version Hardware Version
Description Model number of the IP phone. Serial number of the IP phone. Version number of the IP phone software. Version number of the IP phone hardware.
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Info Tab
B
Description Hardware address of the IP phone. Status of the client certificate, which authenticates the IP phone for use in the ITSP network. This field indicates if the client certificate is properly installed in the IP phone. For an 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. Indicates any additional licenses that you have installed in the IP phone.
Parameter MAC Address Client Certificate
Customization
Licenses
Phone Status
Parameter
Current Time
Description
Current date and time of the system; for example, 10/3/ 2003 16:43:00. Total time elapsed since the last reboot of the system; for example, 25 days and 18:12:36. Total number of broadcast packets sent. Total number of broadcast packets received. Total number of broadcast bytes sent. Total number of broadcast bytes received and processed. Total number of broadcast packets received but not processed. Most codecs can handle up to 5% random packet drops as long as the packets are random and not in groups of two or more. Concurrent packet drops result in voice quality issues. Total number of broadcast bytes received but not processed.
Elapsed Time
Broadcast Pkts Sent Broadcast Bytes Sent Broadcast Pkts Recv Broadcast Bytes Recv
Broadcast Pkts Dropped
Broadcast Bytes Dropped
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Info Tab
B
Description
Total number of RTP packets sent (including redundant packets). Total number of RTP packets received (including redundant packets). Total number of RTP bytes sent. Total number of RTP bytes received. Total number of SIP messages sent (including retransmissions). Total number of SIP messages received (including retransmissions). Total number of bytes of SIP messages sent (including retransmissions). Total number of bytes of SIP messages received (including retransmissions). External IP address used for NAT mapping. ID of the VLAN currently in use if applicable. NOTE Not applicable to WIP310.
Parameter
RTP Packets Sent
RTP Bytes Sent
RTP Packets Recv RTP Bytes Recv SIP Messages Sent
SIP Bytes Sent
SIP Messages Recv
SIP Bytes Recv
External IP Operational VLAN ID
SW Port (Cisco SPA300 Series and Cisco SPA500 Series) PC Port (Cisco SPA303 and Cisco SPA50X)
Displays the type of Ethernet connection from the IP phone to the switch.
Indicates whether the link from the IP phone to a device plugged into the PC port on the phone is up or down.
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Info Tab
Ext Status
The following parameters show for each extension on the phone.
Parameter Registration State
Description Shows Registered if the phone is registered, Not Registered if the phone is not registered to the ITSP. Last date and time the line was registered. Number of seconds before the next registration renewal. Indicates whether the phone user has a new voice mail waiting: Yes or No. This is updated when voice mail notification is received. Port number of the SIP port mapped by NAT.
Last Registration At Next Registration In
Message Waiting
Mapped SIP Port
Line/Call Status
The following parameters show for each line and call on the phone.
Parameter Call State Tone Encoder Decoder Type Remote Hold
Description Status of the call. Type of tone used by the call. Codec used for encoding. Codec used for decoding. Direction of the call. Indicates whether the far end has placed the call on hold. Indicates whether the call was triggered by a call back request. Name of the internal phone.
Callback
Peer Name
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Info Tab
B
Description 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. The port mapped for Real Time Protocol traffic for the call. If the call is a loopback call, displays the loopback mode (source or mirror) and type (media or packet). If the call is not loopback, the field appears blank. Number of milliseconds for decoder latency. Number of milliseconds for receiver jitter. Number of milliseconds for delay in the RTP-to-RTP interface round trip. Number of milliseconds for delay in the internal round trip within the reporting endpoint. Number of packets lost. Number of invalid packets received. The fraction of RTP data packets from the source lost since the beginning of reception. Defined in RFC 3611 RTP Control Protocol Extended Reports (RTCP XR). The fraction of RTP data packets from the source that have been discarded since the beginning of reception, due to late or early arrival, under-run or overflow at the receiving jitter buffer. Defined in RFC 3611RTP Control Protocol Extended Reports (RTCP XR). The mean duration, expressed in milliseconds, of the burst periods that have occurred since the beginning of reception. Defined in RFC 3611RTP Control Protocol Extended Reports (RTCP XR).
Parameter Peer Phone Duration Packets Sent Packets Recv Bytes Sent Bytes Recv Mapped RTP Port
Media Loopback
Decode Latency Jitter Round Trip Delay
End System Delay
Packets Lost Packet Error Loss Rate
Discard Rate
Burst Duration
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Info Tab
B
Description The mean duration, expressed in milliseconds, of the gap periods that have occurred since the beginning of reception. Defined in RFC 3611RTP Control Protocol Extended Reports (RTCP XR). Voice quality metric describing the segment of the call that is carried over this RTP session. Defined in RFC 3611RTP Control Protocol Extended Reports (RTCP XR). The estimated mean opinion score for listening quality (MOS-LQ) is a voice quality metric on a scale from 1 to 5, in which 5 represents excellent and 1 represents unacceptable. Defined in RFC 3611RTP Control Protocol Extended Reports (RTCP XR). The estimated mean opinion score for conversational quality (MOS-CQ) is defined as including the effects of delay and other effects that would affect conversational quality. Defined in RFC 3611RTP Control Protocol Extended Reports (RTCP XR).
Parameter Gap Duration
R Factor
MOS-LQ/MOS Listening
MOS-CQ/MOS Conversational
Downloaded Ring Tone
Parameter Status
Description Indicates whether the phone is downloading a ring tone (and from where) or if it is idle. Information about the user downloaded ring tone 1: name, size, and time-stamp of the tone. Information about the user downloaded ring tone 2: name, size, and time-stamp of the tone.
Ring Tone 1
Ring Tone 2
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System Tab
System Tab
This section describes the fields for the following headings on the System tab: System Configuration, page 233 Internet Connection Type and Static IP Settings, page 235 PPPoE Settings (Cisco SPA525G/525G2 Only), page 236 Optional Network Configuration, page 236 VLAN Settings, page 237 Wi-Fi Settings (Cisco SPA525G/525G2 Only), page 239 Bluetooth Settings (Cisco SPA525G/525G2 Only), page 239 VPN Settings (Cisco SPA525G/525G2 Only), page 239
System Configuration
Parameter Restricted Access Domains (SIP) Enable Web Server
Description This feature is used when implementing software customization. Enable/disable web server of the IP phone. Defaults to yes.
Web Server Port
Port number of the configuration utility. Defaults to 80.
SPA525-http-write (Cisco SPA525G/525G2 SPCP only)
Allow Cisco Configuration Assistant (CCA) or other application to write XML file parameters directly to the phone using HTTP. Choose yes to allow this feature, or no to disable this feature.
Enable Web Admin Access
Lets you enable or disable local access to the configuration utility. Select yes or no from the dropdown menu. Defaults to yes.
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System Tab
B
Description Password for the administrator. Defaults to no password.
Parameter Admin Passwd
User Password
Password for the user. Defaults to no password.
SPA525-protocol (Cisco SPA525G/525G2 only)
Allows you to choose the type of protocol for the phone:
SIPSession Initiation Protocol. Choose if the phone is used with a SIP call control system, such as the Cisco SPA 9000 or a SIP call control system from another provider such as BroadSoft or Asterisk. SPCPSmart Phone Control Protocol. Choose if the phone is used with a Cisco Unified Communications Series server, such as the Cisco Unified Communications 500 Series for Small Business.
SPA525-auto-detect-sccp (Cisco SPA525G/525G2 only)
Choose if the phone should automatically detect the type of protocol used on the network to which it is connected. If set to yes, the phone automatically discovers if it is connected to a call control system using SPCP. If set to yes, the Signaling Protocol and Auto Detect SCCP Settings on the phone are read only. If set to no, the above settings on the phone can be changed by the end user. Allows you to restrict the menus and options that phone users see when they use the phone interface. Choose yes to enable this parameter and restrict access. The default is no. Specific parameters are then designated as na or ro using provisioning files. Parameters designated as na will not appear on the users phone interface. Parameters designated as ro will not be editable by the user.
SPA525-readonly (Cisco SPA525G/525G2 only)
Phone-UI-user-mode
234

System Tab
Internet Connection Type and Static IP Settings
Parameter Internet Connection Type
Description Choose the type of internet connection:

Static IP
DHCP Static IP PPPoE (not applicable to WIP310)
If static IP was chosen as the type of internet connection, displays the static IP address assigned to the phone. If static IP was chosen as the type Default router IP address. Blank if DHCP assigned. LAN Maximum Transmission Unit size. Default value: 1500. Ethernet Maximum Transmission Unit size. Default value: 1500.
Netmask Gateway LAN MTU
Ethernet MTU (Cisco SPA525G//525G2 SPCP only) Duplex Mode
Duplex ModeChoose one of the following to configure the speed/duplex for the phones Ethernet ports:
Auto 10MBps/Duplex 10MBps/Half 100Mbps/Duplex 100MBps/Half
235

System Tab
PPPoE Settings (Cisco SPA525G/525G2 Only)
Parameter PPPoE Login Name
Description Specifies the account name assigned by the ISP for connecting on a Point-to-Point Protocol over Ethernet (PPPoE) link. Specifies the password assigned by the ISP for connecting on a Point-to-Point Protocol over Ethernet (PPPoE) link. Specifies the service name assigned by the ISP for connecting on a Point-to-Point Protocol over Ethernet (PPPoE) link.
PPPoE Login Password
PPPoE Service Name
Optional Network Configuration
Parameter Host Name Domain Primary DNS
Description The host name of the IP phone. The network domain of the IP phone. DNS server used by IP phone in addition to DHCP supplied DNS servers if DHCP is enabled; when DHCP is disabled, this is the primary DNS server. Defaults to 0.0.0.0.
Secondary DNS
DNS server used by IP phone in addition to DHCP supplied DNS servers if DHCP is enabled; when DHCP is disabled, this is the secondary DNS server. Defaults to 0.0.0.0.
DNS Server Order
Specifies the method for selecting the DNS server. The options are Manual, Manual/DHCP, and DHCP/Manual.
236

System Tab
B
Description Do parallel or sequential DNS Query. With parallel DNS query mode, the IP phone sends the same request to all the DNS servers at the same time when doing a DNS lookup, the first incoming reply is accepted by the IP phone. Defaults to parallel.
Parameter DNS Query Mode
Syslog Server
Specify the syslog server name and port. This feature specifies the server for logging IP phone system information and critical events. If both Debug Server and Syslog Server are specified, Syslog messages are also logged to the Debug Server. The debug server name and port. This feature specifies the server for logging IP phone debug information. The level of detailed output depends on the debug level parameter setting. The debug level from 0-3. The higher the level, the more debug information is generated. Zero (0) means no debug information is generated. To log SIP messages, you must set the Debug Level to at least 2. Defaults to 0.
Debug Server
Debug Level
Primary NTP Server Secondary NTP Server Enable Bonjour (SPA525G/ 525G2 only)
IP address or name of primary NTP server. IP address or name of secondary NTP server. Enable Bonjour networking that is used by Office Manager and Cisco Configuration Assistant to discover the Cisco IP phones. Choose yes to enable or no to disable.
VLAN Settings
NOTE Not applicable to the WIP310.
Parameter Enable VLAN
Description Choose Yes to enable VLAN. Choose no to disable.
237

System Tab
B
Description If you use a VLAN without CDP (VLAN enabled and CDP disabled), enter a VLAN ID for the IP phone. Note that only voice packets are tagged with the VLAN ID. Do not use 1 for the VLAN ID. Enables VLAN and priority tagging on the phone data port (802.1p/q). This feature facilities tagging of the VLAN ID (802.1Q) and priority bits (802.1p) of the traffic coming from the PC port of the IP phone. Defaults to No. Choose Yes to enable the tagging algorithm.
Parameter VLAN ID
Enable PC Port VLAN Tagging
PC Port VLAN Highest Priority
0-7 (default 0). The priority applied to all frames, tagged and untagged. The phone modifies the frame priority only if the incoming frame priority is higher than this value. 0-4095 (default 0). Value of the VLAN ID. The phone tags all the untagged frames coming from the PC (it will not tag frames with an existing tag). Enable CDP only if you are using a switch that has Cisco Discovery Protocol. CDP is negotiation based and determines which VLAN the IP phone resides in. Choose yes to enable LLDP-MED for the phone to advertise itself to devices that use that discovery protocol. When the LLDP-MED feature is enabled, after the phone has initialized and Layer 2 connectivity is established, the phone sends out LLDP-MED PDU frames. If the phone receives no acknowledgment, the manually configured VLAN or default VLAN will be used if applicable. If the CDP is used concurrently, the waiting period of 6 seconds is used. The waiting period will increase the overall startup time for the phone.
PC Port VLAN ID
Enable CDP
Enable LLDP-MED
238

System Tab
B
Description
Parameter Network Startup Delay
Setting this value causes a delay for the switch to get to the forwarding state before the phone will send out the first LLDP-MED packet. The default delay is 3 seconds. For configuration of some switches, you might need to increase this value to a higher value for LLDP-MED to work. Configuring a delay can be important for networks that use Spanning Tree Protocol.
Wi-Fi Settings (Cisco SPA525G/525G2 Only)
Parameter
SPA525-wifi-on
Description Set to yes to enable Wireless-G service on the Cisco SPA525G/525G2.
Bluetooth Settings (Cisco SPA525G/525G2 Only)
Parameter
Enable BT
Description
Set to yes to enable support for Bluetooth devices on the Cisco SPA525G.
VPN Settings (Cisco SPA525G/525G2 Only)
Parameter
VPN Server
Description
The IP address of the VPN server to which the phone connects. Username configured on the VPN server for the phone. Password associated with the username configured on the VPN for the phone.
VPN User Name
VPN Password
239

SIP Tab
B
Description
(Optional) The tunnel group, if required by the VPN server. If the phone should attempt to connect to the VPN each time it is powered on. Choose yes to have the phone try to automatically connect, or no to keep the default behavior.
Parameter
VPN Tunnel Group
Connect on Bootup
SIP Tab
This section describes the fields for the following headings on the SIP tab: SIP Parameters, page 240 SIP Timer Values (sec), page 244 Response Status Code Handling, page 246 RTP Parameters, page 247 SDP Payload Types, page 249 NAT Support Parameters, page 252 Linksys Key System Parameters, page 254
SIP Parameters
Parameter Max Forward
Description SIP Max Forward value, which can range from 1 to 255. Defaults to 70.
Max Redirection
Number of times an invite can be redirected to avoid an infinite loop. Defaults to 5.
240

SIP Tab
B
Description Maximum number of times (from 0 to 255) a request may be challenged. Defaults to 2.
Parameter Max Auth
SIP User Agent Name
Used in outbound REGISTER requests. Defaults to $VERSION. If empty, the header is not included. Macro expansion of $A to $D corresponding to GPP_A to GPP_D allowed.
SIP Server Name
Server header used in responses to inbound responses. Defaults to $VERSION.
SIP Reg User Agent Name
User-Agent name to be used in a REGISTER request. If this is not specified, the <SIP User Agent Name> is also used for the REGISTER request. Defaults to blank.
SIP Accept Language
Accept-Language header used. To access, click the SIP tab, and fill in the SIP Accept Language field. There is no default (this indicates IP phone does not include this header). If empty, the header is not included.
DTMF Relay MIME Type
MIME Type used in a SIP INFO message to signal a DTMF event. This field must match that of the Service Provider. Defaults to application/dtmf-relay.
Hook Flash MIME Type
MIME Type used in a SIP INFO message to signal a hook flash event. The default is application/hook-flash.
Remove Last Reg
Lets you remove the last registration before registering a new one if the value is different. Select yes or no from the drop-down menu. Defaults to no.
241

SIP Tab
B
Description Lets you use compact SIP headers in outbound SIP messages. Select yes or no from the drop-down menu. If set to yes, the phone uses compact SIP headers in outbound SIP messages. If set to no, the phone uses normal SIP headers. If inbound SIP requests contain compact headers, the phone reuses the same compact headers when generating the response regardless the settings of the <Use Compact Header> parameter. If inbound SIP requests contain normal headers, the phone substitutes those headers with compact headers (if defined by RFC 261) if <Use Compact Header> parameter is set to yes. Default: no
Parameter Use Compact Header
Escape Display Name
Lets you keep the Display Name private. Select yes if you want the IP phone 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. Defaults to yes.
SIP-B Enable
Enables SIP for Business (supports Sylantro call flows) call features. Enables support for the BroadSoft Talk Package, which enables a user to answer or resume a call by clicking a button in an external application. Enables support for the BroadSoft Hold Package, which enables a user to place a call on hold by clicking a button in an external application. Enables support for the BroadSoft Conference Package, which enables a user to start a conference by clicking a button in an external application. If enabled, the unit will send out a NOTIFY with event=conference when starting a conference. If set to yes, unit will include c=0.0.0.0 syntax in SDP when sending a SIP re-INVITE to the peer to hold the call. If set to no, unit will not include the c=0.0.0.0 syntax in the SDP. The unit will always include a=sendonly syntax in the SDP in either case. Defaults to yes.
Talk Package
Hold Package
Conference Package
Notify Conference
RFC 2543 Call Hold
242

SIP Tab
B
Description If set to yes, the Cisco IP phone uses a different random call-ID for registration after the next software reboot. If set to no, the Cisco IP phone tries to use the same call-ID for registration after the next software reboot. The Cisco IP phone always uses a new random Call-ID for registration after a power-cycle, regardless of this setting. Defaults to no.
Parameter Random REG CID On Reboot
Mark All AVT packets
If set to yes, all audio video transport (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. Defaults to yes.
SIP TCP Port Min
Specifies the lowest TCP port number that can be used for SIP sessions. Defaults to 5060. Specifies the highest TCP port number that can be used for SIP sessions. Defaults to 5080. The CTI interface allows a third-party application to control and monitor the state of a phone that has registered with the Cisco SPA 9000. With this interface, an application can control a phone to initiate an outgoing call or answer an incoming call with a mouse click from a PC. Provides the option to take the caller ID from PAIDRPID-FROM, P-ASSERTEDIDENTITY, REMOTE-PARTYID, or FROM header. Selects the method to use for SRTP. Two choices are available:
SIP TCP Port Max
CTI Enable
Caller ID Header
SRTP Method
x-sipuralegacy SRPT method s-descriptornew method compliant with RFC3711 and RFC-4568
The default value is "x-sipura. NOTE Not applicable to WIP310.
243

SIP Tab
B
Description Controls whether to hold call leg with transfer target before sending REFER to the transferee when initiating a fully-attended call transfer (where the transfer target has answered). Default value is "no, where the call leg is not held. NOTE Not applicable to WIP310.
Parameter Hold Target Before REFER
SIP Timer Values (sec)
Parameter SIP T1
Description RFC 3261 T1 value (RTT estimate), which can range from 0 to 64 seconds. Defaults to .5 seconds. RFC 3261 T2 value (maximum retransmit interval for non-INVITE requests and INVITE responses), which can range from 0 to 64 seconds. Defaults to 4 seconds. RFC 3261 T4 value (maximum duration a message remains in the network), which can range from 0 to 64 seconds. Defaults to 5 seconds. INVITE time-out value, which can range from 0 to 64 seconds. Defaults to 16 seconds. Non-INVITE time-out value, which can range from 0 to 64 seconds.Defaults to 16 seconds. INVITE final response, time-out value, which can range from 0 to 64 seconds.Defaults to 16 seconds. ACK hang-around time, which can range from 0 to 64 seconds. Defaults to 16 seconds. Non-INVITE response hang-around time, which can range from 0 to 64 seconds. Defaults to 16 seconds. INVITE request Expires header value. If you enter 0, the Expires header is not included in the request. Ranges from 0 to 19999999999999999999999999999999. Defaults to 240 seconds.
SIP T2
SIP T4
SIP Timer B
SIP Timer F
SIP Timer H
SIP Timer D
SIP Timer J
INVITE Expires
244

SIP Tab
B
Description ReINVITE request Expires header value. If you enter 0, the Expires header is not included in the request. Ranges from 0 to 19999999999999999999999999999999. Defaults to 30
Parameter ReINVITE Expires
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. Defaults to 1 second.
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. Defaults to 7200.
Reg Retry Intvl (See note below)
Interval to wait before the IP phone retries registration after failing during the last registration. Defaults to 30.
Reg Retry Long Intvl (See note below)
When registration fails with a SIP response code that does not match<Retry Reg RSC>, the IP phone waits for the specified length of time before retrying. If this interval is 0, the IP phone stops trying. This value should be much larger than the Reg Retry Intvl value, which should not be 0. Defaults to 1200.
Reg Retry Random Delay
Random delay range (in seconds) to add to <Register Retry Intvl> when retrying REGISTER after a failure. This feature was added in Release 5.1. Defaults to blank, which disables this feature.
Reg Retry Long Random Delay
Random delay range (in seconds) to add to <Register Retry Long Intvl> when retrying REGSITER after a failure. This feature was added in Release 5.1. Defaults to blank, which disables this feature.
245

SIP Tab
B
Description The maximum value to cap the exponential back-off retry delay (which starts at <Register Retry Intvl> and doubles on every REGISTER retry after a failure). In other words, the retry interval is always at <Register Retry Intvl> seconds after a failure. If this feature is enabled, <Reg Retry Random Delay> is added on top of the exponential back-off adjusted delay value. This feature was added in Release 5.1. Defaults to blank, which disables the exponential backoff feature.
Parameter Reg Retry Intvl Cap
Sub Min Expires
This value sets the lower limit of the REGISTER expires value returned from the Proxy server. This value sets the upper limit of the REGISTER minexpires value returned from the Proxy server in the Min-Expires header. Defaults to 7200. This value (in seconds) determines the retry interval when the last Subscribe request fails. Defaults to 10.
Sub Max Expires
Sub Retry Intvl
NOTE Cisco IP phones can use a RETRY-AFTER value when received from a SIP proxy
server that is too busy to process a request (503 Service Unavailable message). If the response message includes a RETRY-AFTER header, the phone waits for the specified length of time before retrying to REGISTER again. If a RETRY-AFTER header is not present, the phone waits for the value specified in the Reg Retry Interval or the Reg Retry Long Interval parameter.
Response Status Code Handling
Parameter SIT1 RSC
Description 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. Defaults to blank.
246

SIP Tab
B
Description SIP response status code to INVITE on which to play the SIT2 Tone. Defaults to blank. SIP response status code to INVITE on which to play the SIT3 Tone. Defaults to blank. SIP response status code to INVITE on which to play the SIT4 Tone. Defaults to blank. SIP response code that retries a backup server for the current request. Defaults to blank. Interval to wait before the IP phone retries registration after failing during the last registration. Defaults to blank.
Parameter SIT2 RSC
SIT3 RSC
SIT4 RSC
Try Backup RSC
Retry Reg RSC
RTP Parameters
Parameter RTP Port Min
Description Minimum port number for RTP transmission and reception. Minimum port number for RTP transmission and reception. Should define a range that contains at least 10 even number ports (twice the number of lines); for example, configure RTP port min to 16384 and RTP port max to 16402. Defaults to 16384.
RTP Port Max
Maximum port number for RTP transmission and reception. Should define a range that contains at least 10 even number ports (twice the number of lines); for example, configure RTP port min to 16384 and RTP port max to 16402. Defaults to 16482.
247

SIP Tab
B
Description Packet size in seconds, which can range from 0.01 to 0.16. Valid values must be a multiple of 0.01 seconds. Defaults to 0.030.
Parameter RTP Packet Size
Max RTP ICMP Err
Number of successive ICMP errors allowed when transmitting RTP packets to the peer before the IP phone terminates the call. If value is set to 0, the IP phone ignores the limit on ICMP errors. Defaults to 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 IP phone 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/IP phone-1.0.31(b)). The NTP timestamp used in the SR is a snapshot of the IP phones local time, not the time reported by an NTP server. If the IP phone 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 IP phone web page. Defaults to 0.
No UDP Checksum
Select yes if you want the IP phone to calculate the UDP header checksum for SIP messages. Otherwise, select no. Defaults to no.
Symmetric RTP
Enable symmetric RTP operation. If enabled, sends RTP packets to the source address and port of the last received valid inbound RTP packet. If disabled (or before the first RTP packet arrives) sends RTP to the destination as indicated in the inbound SDP. Defaults to no.
248

SIP Tab
B
Description Determines whether the IP phone includes the P-RTPStat 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>. Defaults to no.
Parameter Stats In BYE
SDP Payload Types

The configured dynamic payloads are used for outbound calls only where the IP phone presents the SDP offer. For inbound calls with a SDP offer, the IP phone follows the caller dynamic payload type assignments. The IP phone uses the configured codec names in its outbound SDP. The IP phone ignores the codec names in incoming SDP for standard payload types (0 95). For dynamic payload types, the IP phone identifies the codec by the configured codec names. Comparison is case-insensitive.
Parameter AVT Dynamic Payload
Description AVT dynamic payload type. Ranges from 96-127. Defaults to 101.
INFOREQ Dynamic Payload
INFOREQ dynamic payload type. Defaults to blank.
G.726-16 dynamic payload type. Ranges from 96-127. Defaults to 98. NOTE Not applicable to Cisco SPA525G/WIP310.
249

SIP Tab
B
Description G.726-24 dynamic payload type. Ranges from 96-127. Defaults to 97. NOTE Not applicable to Cisco SPA525G/525G2 and WIP310.
Parameter G726r24 Dynamic Payload
G726r32 dynamic payload type. The default is 2.
G.726-40 dynamic payload type. Ranges from 96-127. Defaults to 96. NOTE Not applicable to Cisco SPA525G/525G2 and WIP310.
G729b Dynamic Payload
G729b Dynamic Payload type. Defaults to 99.
EncapRTP Dynamic Payload
EncapRTP Dynamic Payload type. Defaults to 112.
RTP-Start-Loopback Dynamic
RTP-Start-Loopback Dynamic Payload. Defaults to 113.
RTP-Start-Loopback Codec
RTP-Start-Loopback Codec. Select one of following: G711u, G711a, G726-16, G726-24, G726-32, G726-40, G729a, or G723. Cisco SPA525G choices: G711u, G711a, G726-32, G729a, G722. Defaults to G711u.
AVT Codec Name
AVT codec name used in SDP. Defaults to telephone-event.
G711u Codec Name
G.711u codec name used in SDP. Defaults to PCMU.
G711a Codec Name
G.711a codec name used in SDP. Defaults to PCMA.
250

SIP Tab
B
Description G.726-16 codec name used in SDP. Defaults to G726-16. NOTE Not applicable to Cisco SPA525G/525G2 and WIP310.
Parameter G726r16 Codec Name
G726r24 Codec Name
G.726-24 codec name used in SDP. Defaults to G726-24. NOTE Not applicable to Cisco SPA525G/525G2 and WIP310.
G726r32 Codec Name
G.726-32 codec name used in SDP. Defaults to G726-32.
G726r40 Codec Name
G.726-40 codec name used in SDP. Defaults to G726-40. NOTE Not applicable to Cisco SPA525G/525G2 and WIP310.
G729a Codec Name
G.729a codec name used in SDP. Defaults to G729a.
G729b Codec Name
G.729b codec name used in SDP. Defaults to G729ab.
G722 Codec Name
G.722 codec name used in SDP. Defaults to G722. NOTE Not supported on the WIP310.
EncapRTP Codec Name
EncapRTP codec name used in SDP. Defaults to encaprtp.
251

SIP Tab
NAT Support Parameters
Parameter Handle VIA received
Description If you select yes, the phone processes the received parameter in the VIA header (this is inserted by the server in a response to any of its requests). If you select no, the parameter is ignored. Select yes or no from the drop-down menu. Defaults to no.
Handle VIA rport
If you select yes, the IP phone processes the rport parameter in the VIA header (this is inserted by the server in a response to any of its requests). If you select no, the parameter is ignored. Select yes or no from the drop-down menu. Defaults to no.
Insert VIA received
Inserts the received parameter into the VIA header of SIP responses if the received-from IP and VIA sent-by IP values differ. Select yes or no from the drop-down menu. Defaults to no.
Insert VIA rport
Inserts the rport 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. Defaults to 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. Defaults to no.
Send Resp To Src Port
Sends responses to the request source port instead of the VIA sent-by port. Select yes or no from the dropdown menu. Defaults to no.
STUN Enable
Enables the use of STUN to discover NAT mapping. Select yes or no from the drop-down menu. Defaults to no.
252

SIP Tab
B
Description If the STUN Enable feature is enabled and a valid STUN server is available, the IP phone can perform a NATtype 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 IP phone detects symmetric NAT or a symmetric firewall, NAT mapping is disabled. Defaults to no.
Parameter STUN Test Enable
STUN Server
IP address or fully-qualified domain name of the STUN server to contact for NAT mapping discovery. You can use a public STUN server or set up your own STUN server. External IP address to substitute for the actual IP address of the IP phone in all outgoing SIP messages. If 0.0.0.0 is specified, no IP address substitution is performed. If this parameter is specified, the IP phone 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. Defaults to blank.
EXT IP
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. Defaults to blank.
NAT Keep Alive Intvl
Interval between NAT-mapping keep alive messages. Defaults to 15.
253

Provisioning Tab
Linksys Key System Parameters
Parameter Linksys Key System
Description Enable or disable the Linksys Key System on the IP phone. Defaults to yes.
Multicast Address
The multicast address is used by the Cisco SPA 9000 to communicate with the Cisco SPA IP phones. Defaults to 224.168.168.168:6061.
Key System Auto Discovery
Enables or disables auto discovery of the call control server (for example, the Cisco SPA 9000). Disable this feature for teleworkers or other scenarios where multicast does not work. IP address of the call control server IP. Enter the IP address for teleworkers or other scenarios where multicast does not work. The choices are: none, G.711u, or G.711a. Defaults to none.
Key System IP Address
Force LAN Codec
Provisioning Tab
NOTE For information about the Provisioning page, see the Cisco Small Business IP
254

Regional Tab
Regional Tab
This section describes the fields for the following headings on the Regional tab: Call Progress Tones, page 255 Distinctive Ring Patterns, page 258 Control Timer Values (sec), page 259 Vertical Service Activation Codes, page 260 Vertical Service Announcement Codes, page 265 Outbound Call Codec Selection Codes, page 265 Time (Cisco SPA525G/525G2 Only), page 268 Language (Cisco SPA525G/525G2 Only), page 268 Miscellaneous, page 268
Call Progress Tones
Parameter Dial Tone
Description Prompts the user to enter a phone number. Defaults to 350@-19,440@-19;10(*/0/1+2).
Bluetooth Dial Tone (Cisco SPA525G/525G2 only)
Tone that indicates a bluetooth headset is on and the user can make a call. Defaults to 350@-19,440@-19;1(0/*/0);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. Defaults to 420@-16;10(*/0/1).
Prompt Tone
Prompts the user to enter a call forwarding phone number. Defaults to 520@-19,620@-19;10(*/0/1+2).
255

Regional Tab
B
Description Played when a 486 RSC is received for an outbound call. Defaults to 480@-19,620@-19;10(.5/.5/1+2).
Parameter Busy Tone
Reorder Tone
Played when an outbound call has failed or after the far end hangs up during an established call. Reorder Tone is played automatically when <Dial Tone> or any of its alternatives times out. Defaults to 480@-19,620@-19;10(.25/.25/1+2).
Off Hook Warning Tone
Played when the caller has not properly placed the handset on the cradle. Off Hook Warning Tone is played when Reorder Tone times out. Defaults to 480@10,620@0;10(.125/.125/1+2).
Ring Back Tone
Played during an outbound call when the far end is ringing. Defaults to 440@-19,480@-19;*(2/4/1+2).
Call Waiting Tone
Played when a call is waiting. Defaults to 440@10;30(.3/9.7/1) Brief tone to notify the user that the last input value has been accepted. Defaults to 600@-16; 1(.25/.25/1).
Confirm Tone
SIT1 Tone
Alternative to the Reorder Tone played when an error occurs as a caller makes an outbound call. The RSC to trigger this tone is configurable on the SIP screen. Defaults to 985@-16,1428@-16,1777@-16;20(.380/0/ 1,.380/0/2,.380/0/3,0/4/0).
SIT2 Tone
Alternative to the Reorder Tone played when an error occurs as a caller makes an outbound call. The RSC to trigger this tone is configurable on the SIP screen. Defaults to 914@-16,1371@-16,1777@-16;20(.274/0/ 1,.274/0/2,.380/0/3,0/4/0).
256

Regional Tab
B
Description Alternative to the Reorder Tone played when an error occurs as a caller makes an outbound call. The RSC to trigger this tone is configurable on the SIP screen. Defaults to 914@-16,1371@-16,1777@-16;20(.380/0/ 1,.380/0/2,.380/0/3,0/4/0)
Parameter SIT3 Tone
SIT4 Tone
This is an alternative to the Reorder Tone played when an error occurs as a caller makes an outbound call. The RSC to trigger this tone is configurable on the SIP screen. Defaults to 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 callers mailbox. Defaults to 350@-19,440@-19;2(.1/.1/1+2);10(*/0/1+2).
Cfwd Dial Tone
Played when all calls are forwarded. Defaults to 350@-19,440@-19;2(.2/.2/1+2);10(*/0/1+2).
Holding Tone
Informs the local caller that the far end has placed the call on hold. Defaults to 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. Defaults to 350@-19;20(.1/.1/1,.1/9.7/1).
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. Defaults to 397@-19,507@-19;15(0/2/0,.2/.1/1,.1/2.1/2).
Page Tone
Specifies the tone transmitted when the paging feature is enabled. Defaults to 600@-16;.3(.05/0.05/1).
Alert Tone
Played when an alert occurs. Defaults to 600@-19;.2(.05/0.05/1).
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B
Description Audible notification tone played when a system error occurs. Defaults to 600@-16;.1(.05/0.05/1). NOTE WIP310 and Cisco SPA525G/525G2 only.
Parameter System Beep
Distinctive Ring Patterns
Parameter Cadence 1
Description Cadence script for distinctive ring 1. Defaults to 60(2/4).
Cadence 2
Cadence script for distinctive ring 2. Defaults to 60(.3/.2, 1/.2,.3/4.
Cadence 3
Cadence script for distinctive ring 3. Defaults to 60(.8/.4,.8/4).
Cadence 4
Cadence script for distinctive ring 4. Defaults to 60(.4/.2,.3/.2,.8/4).
Cadence 5
Cadence script for distinctive ring 5. Defaults to 60(.2/.2,.2/.2,.2/.2,1/4)
Cadence 6
Cadence script for distinctive ring 6. Defaults to60(.2/.4,.2/.4,.2/4).
Cadence 7
Cadence script for distinctive ring 7. Defaults to 60(4.5/4).
Cadence 8
Cadence script for distinctive ring 8. Defaults to 60(0.25/9.75)
Cadence 9
Cadence script for distinctive ring 9. Defaults to 60(.4/.2,.4/2).
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Parameter Reorder Delay
Description Delay after far end hangs up before reorder (busy) tone is played. 0 = plays immediately, inf = never plays. Range: 0255 seconds. Set to 255 to return the phone immediately to on-hook status and to not play the tone. Defaults to 5.
Call Back Expires
Expiration time in seconds of a call back activation. Range: 065535 seconds. Defaults to 1800.
Call Back Retry Intvl
Call back retry interval in seconds. Range: 0255 seconds. Defaults to 30.
Call Back Delay
Delay after receiving the first SIP 18x response before declaring the remote end is ringing. If a busy response is received during this time, the IP phone still considers the call as failed and keeps on retrying. Defaults to 0.5.
Interdigit Long Timer
Long timeout between entering digits when dialing. The interdigit timer values are used as defaults when dialing. The Interdigit_Long_Timer is used after any one digit, if all valid matching sequences in the dial plan are incomplete as dialed. Range: 064 seconds. Defaults to 10.
Interdigit Short Timer
Short timeout between entering digits when dialing. The Interdigit_Short_Timer is used after any one digit, if at least one matching sequence is complete as dialed, but more dialed digits would match other as yet incomplete sequences. Range: 064 seconds. Defaults to 3.
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Vertical Service Activation Codes

The following Vertical Service Activation Codes are automatically appended to the dial plan.
Parameter Call Return Code
Description This code calls the last caller. Defaults to *69.
Blind Transfer Code
Begins a blind transfer of the current call to the extension specified after the activation code. Defaults to *98.
Call Back Act Code
Starts a callback when the last outbound call is not busy. Defaults to *66.
Call Back Deact Code
Cancels a callback. Defaults to *86.
Cfwd All Act Code
Forwards all calls to the extension specified after the activation code. Defaults to *72.
Cfwd All Deact Code
Cancels call forwarding of all calls. Defaults to *73.
Cfwd Busy Act Code
Forwards busy calls to the extension specified after the activation code. Defaults to *90.
Cfwd Busy Deact Code
Cancels call forwarding of busy calls. Defaults to *91.
Cfwd No Ans Act Code
Forwards no-answer calls to the extension specified after the activation code. Defaults to *92.
Cfwd No Ans Deact Code
Cancels call forwarding of no-answer calls. Defaults to *93.
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B
Description Enables call waiting on all calls. Defaults to *56.
Parameter CW Act Code
CW Deact Code
Disables call waiting on all calls. Defaults to *57.
CW Per Call Act Code
Enables call waiting for the next call. Defaults to *71.
CW Per Call Deact Code
Disables call waiting for the next call. Defaults to *70.
Block CID Act Code
Blocks caller ID on all outbound calls. Defaults to *67.
Block CID Deact Code
Removes caller ID blocking on all outbound calls. Defaults to *68.
Block CID Per Call Act Code
Blocks caller ID on the next outbound call. Defaults to *81.
Block CID Per Call Deact Code Block ANC Act Code
Removes caller ID blocking on the next inbound call. Defaults to *82. Blocks all anonymous calls. Defaults to *77.
Block ANC Deact Code
Removes blocking of all anonymous calls. Defaults to *87.
DND Act Code
Enables the do not disturb feature. Defaults to *78.
DND Deact Code
Disables the do not disturb feature. Defaults to *79.
Secure All Call Act Code
Makes all outbound calls secure. Defaults to *16.
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B
Description Makes all outbound calls not secure. Defaults to *17.
Parameter Secure No Call Act Code
Secure One Call Act Code
Makes the next outbound call secure. (It is redundant if all outbound calls are secure by default.) Defaults to *18.
Secure One Call Deact Code
Makes the next outbound call not secure. (It is redundant if all outbound calls are not secure by default.) Defaults to *19.
Paging Code
The star code used for paging the other clients in the group. Defaults to *96.
Call Park Code
The star code used for parking the current call. Defaults to *38.
Call Pickup Code
The star code used for picking up a ringing call. Defaults to *36.
Call UnPark Code
The star code used for picking up a call from the call park. Defaults to *39.
Group Call Pickup Code
The star code used for picking up a group call. Defaults to *37.
Media Loopback Code
The star code used for media loopback. Defaults to *03.
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B
Description These codes tell the IP phone what to do when the user places the current call on hold and is listening to the second dial tone. One or more *code can be configured into this parameter, such as *98, or *97|*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 IP phone to perform a blind transfer to a target number that is prepended by the service *code. For example, after the user dials *98, the IP phone 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 IP phone sends a blind REFER to the holding party with the Refer-To target equals to *98<target_number>. This feature allows the IP phone to hand off a call to an application server to perform further processing, such as call park. The *codes should not conflict with any of the other vertical service codes internally processed by the IP phone. You can empty the corresponding *code that you do not want to IP phone to process.
Parameter Referral Services Codes
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B
Description These codes tell the IP phone 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 IP phone to call the target number prepended by the *code. For example, after user dials *72, the IP phone plays a prompt tone awaiting the user to enter a valid target number. When a complete number is entered, the IP phone sends a INVITE to *72<target_number> as in a normal call. This feature allows the proxy to process features like call forward (*72) or BLock Caller ID (*67). The *codes should not conflict with any of the other vertical service codes internally processed by the IP phone. You can empty the corresponding *code that you do not want to IP phone 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 *72c|*67p. Below are a list of allowed tone parameters (note the use of back quotes surrounding the parameter w/o spaces)
Parameter Feature Dial Services Codes
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 IP phone 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 IP phone send INVITE *73@..... as usual when user dials *73.
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Vertical Service Announcement Codes

Service Annc (Announcement) Base Number: Defaults to blank. Service Annc (Announcement) Extension Codes: Defaults to blank.
Outbound Call Codec Selection Codes

These codes automatically appended to the dial plan. You do not need to include them in the dial plan.
Parameter Prefer G711u Code
Description Makes this codec the preferred codec for the associated call. Defaults to *017110.
Force G711u Code
Makes this codec the only codec that can be used for the associated call. Defaults to *027110.
Prefer G711a Code
Makes this codec the preferred codec for the associated call. Defaults to *017111
Force G711a Code
Prefer G722 Code
Makes this codec the preferred codec for the associated call. Defaults to *01722. Only one G.722 call at a time is allowed. If a conference call is placed, a SIP re-invite message is sent to switch the calls to narrowband audio. NOTE Not supported on the WIP310.
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B
Description Makes this codec the only codec that can be used for the associated call. Defaults to *02722. Only one G.722 call at a time is allowed. If a conference call is placed, a SIP re-invite message is sent to switch the calls to narrowband audio. NOTE Not supported on the WIP310.
Parameter Force G722 Code
Prefer L16 Code
Force L 16 Code
Prefer G723 Code
Makes this codec the preferred codec for the associated call. Defaults to *01723. NOTE Not applicable to WIP310 or Cisco SPA525G.
Force G723 Code
Makes this codec the only codec that can be used for the associated call. Defaults to *02723. NOTE Not applicable to WIP310 or Cisco SPA525G/ 525G2.
Prefer G726r16 Code
Makes this codec the preferred codec for the associated call. Defaults to *0172616. NOTE Not applicable to WIP310 or Cisco SPA525G/ 525G2.
Force G726r16 Code
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B
Description Makes this codec the preferred codec for the associated call. Defaults to *0172624. NOTE Not applicable to WIP310 or Cisco SPA525G/ 525G2.
Parameter Prefer G726r24 Code
Force G726r24 Code
Prefer G726r32 Code
Makes this codec the preferred codec for the associated call. Defaults to *0172632.
Force G726r32 Code
Prefer G726r40 Code
Makes this codec the preferred codec for the associated call. Defaults to *0172640. NOTE Not applicable to WIP310 or Cisco SPA525G/ 525G2.
Force G726r40 Code
Prefer G729a Code
Makes this codec the preferred codec for the associated call. Defaults to *01729.
Force G729a Code
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Time (Cisco SPA525G/525G2 Only)
Parameter
Time Zone
Description
Selects the number of hours to add to GMT to generate the local time for caller ID generation. Choices are GMT-12:00, GMT-11:00,, GMT, GMT+01:00, GMT+02:00, , GMT+13:00. Defaults to GMT-08:00.
Time Offset
This specifies the offset from GMT to use for the local system time. See Daylight Saving Time Rule in Miscellaneous, page 268. Select yes to enable Daylight Saving Time.
Daylight Saving Time Rule
Daylight Saving Time Enable
Language (Cisco SPA525G/525G2 Only)
Parameter
Dictionary Server Script.
Description
See Dictionary Server Script in Miscellaneous, page 268. See Language Selection in Miscellaneous, page 268.
Language Selection
Miscellaneous
Parameter Set Local Date (mm/dd)
Description Sets the local date (mm represents the month and dd represents the day). The year is optional and uses two or four digits. NOTE Not applicable to the Cisco SPA525G/525G2.
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B
Description Sets the local time (hh represents hours and mm represents minutes). Seconds are optional. NOTE Not applicable to the Cisco SPA525G/525G2.
Parameter Set Local Time (HH/mm)
Time Zone
Selects the number of hours to add to GMT to generate the local time for caller ID generation. Choices are GMT-12:00, GMT-11:00,, GMT, GMT+01:00, GMT+02:00, , GMT+13:00. Defaults to GMT-08:00. NOTE Found in the Time section for the Cisco SPA525G/525G2.
Time Offset (HH/mm)
This specifies the offset from GMT to use for the local system time. NOTE Found in the Time section for the Cisco SPA525G/525G2.
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B
Description 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. This is the format of the rule: 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 131. 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).
Parameter Daylight Saving Time Rule
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B
Description The <weekday> value equals any value in the range 17 (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. NOTE Found in the Time section for the Cisco SPA525G/525G2.
Parameter
Daylight Saving Time Enable
Select yes to enable Daylight Saving Time. NOTE Found in the Time section for the Cisco SPA525G/525G2.
DTMF Playback Level
Local DTMF playback level in dBm, up to one decimal place. Defaults to -16.
DTMF Playback Length
Local DTMF playback duration in milliseconds. Defaults to .1.
Inband DTMF Boost
Controls the amount of amplification applied DTMF signals. Choices are OdB, 3dB, 6dB, 9dB, 12dB, 15dB, or 18dB. Defaults to 12dB.
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B
Description Defines the location of the dictionary server, the languages available and the associated dictionary. The syntax is as follows:
Parameter Dictionary Server Script/ SCCP Dictionary Server Script (Cisco SPA525G/ 525G2 SCCP only)
<Dictionary_Server_Script ua="na"> </Dictionary_Server_Script>

Defaults to blank and the maximum number of characters is 512. The detailed format is as follows:
serv={server ip port and root path}; d0=<language0>;x0=<dictionary0 filename>; d1=<language1>;x1=<dictionary1 filename>; d2=<language2>;x2=<dictionary2 filename>; d3=<language3>;x3=<dictionary3 filename>; d4=<language4>;x4=<dictionary4 filename>; d5=<language5>;x5=<dictionary5 filename>; d6=<language6>;x6=<dictionary6 filename>; d7=<language3>;x7=<dictionary7 filename>; d8=<language8>;x8=<dictionary8 filename>; d9=<language5>;x9=<dictionary9 filename>;
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B
Description The following is an example value:
Parameter
<Dictionary_Server_Script ua="na"> serv=tftp://192.168.1.119/ ;d0=English;x0=enS_v101.xml;d1=Spanish ;x1=esS_v101.xml </ Dictionary_Server_Script>

NOTE Language Selection/SCCP Language Selection (Cisco SPA525G/525G2 SCCP Only) Not applicable to the WIP310.
Specifies the default language. The value needs to match one of the languages supported by the dictionary server. The script (dx value) is as follows:
<Language_Selection ua="na"> </Language_Selection>

Defaults to blank and the maximum number of characters is 512. The following is an example:
<Language_Selection ua="na"> Spanish </Language_Selection>

NOTE Default Character Encoding (Cisco SPA303, Cisco SPA50X Not applicable to the WIP310.
The default is ISO-8859-1 for backward compatibility with Cisco SPA900 series phones. If set to UTF-8, line keys and other labels entered via the configuration utility containing UTF-8 characters will be displayed correctly on the phone. (SIP only)
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Phone Tab
This section describes the fields for the following headings on the Phone tab: General, page 274 Line Key, page 277 Miscellaneous Line Key Settings, page 279 Line Key LED Pattern, page 279 Supplementary Services, page 281 Ring Tone (Cisco SPA300 Series and Cisco SPA500 Series), page 283 Ring Tone (WIP310), page 284 Auto Input Gain (dB), page 285 Extension Mobility, page 291 XML Service, page 291 Lightweight Directory Access Protocol (LDAP) Corporate Directory Search, page 288 Programmable Softkeys, page 292
General
Parameter Station Display Name
Description Name to identify the station; appears on the LCD screen on phone models that have a display. You can use spaces in this field and the name does not have to be unique. If both the Station Display Name and Station Name fields are populated, the Station Display Name field takes precedence and is displayed on the phone. Name to identify this station; appears on the LCD screen on phone models that have a display. No spaces are allowed and the name must be unique.
Station Name
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B
Description Phone number or URL to check voice mail. The service provider often hosts a voice mail service. The advantages of hosted voice mail include:
Parameter Voice Mail Number

Text Logo
Advanced features such as voice mail to email conversion. Calls can go to voice mail when the broadband connection is down.
Text logo to display when the phone boots up. A service provider, for example, can enter logo text as follows:
Up to 2 lines of text Each line must be fewer than 32 characters Insert a new line character (\n) between lines Insert escape code %0a
For example, Super\n%0aTelecom will display:
Super Telecom
Use the + character to add spaces for formatting. For example, you can add multiple + characters before and after the text to center it. NOTE Not applicable to the WIP310, SPA301, or SPA501. On the Cisco SPA525G/525G2, this setting is located in the User tab. See Screen (Cisco SPA525G/ 525G2), page 314. BMP Picture Download URL URL locating the bitmap (.BMP) file to display on the LCD background. For more information, see the Configuring Phone Information and Display Settings section on page 46. NOTE Not applicable to the WIP310 or SPA301. On the Cisco SPA525G/525G2, this setting is located in the User tab. See Screen (Cisco SPA525G/525G2), page 314.
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B
Description Select from Default, BMP Picture, Text Logo, or None. Defaults to Default. NOTE Not applicable to the WIP310, SPA301, or SPA501. On the Cisco SPA525G/525G2, this setting is located in the User tab. See Screen (Cisco SPA525G/ 525G2), page 314.
Parameter Select Logo
Select Background Picture
Select from Default, BMP Picture, or None. Defaults to Default. NOTE Not applicable to the WIP310, SPA301, or SPA501. On the Cisco SPA525G/525G2, this setting is located in the User tab. See Screen (Cisco SPA525G/ 525G2), page 314.
Softkey Labels Font
Choose the font width for the softkey labels to display on your phone. See Customizing Phone Softkeys, page 62. NOTE Not applicable to the WIP310, SPA301, or SPA501.
Screen Saver Enable
Enables a screen saver on the phones LCD. When the phone is idle for a specified time, it enters screen saver mode. (Users can set up screen savers directly using phone Setup button.) Any button press or on/off hook event triggers the phone to return to its normal mode. (The screen shows Press any key to unlock your phone.) If a user password is set, the user must enter it to exit screen saver mode. NOTE Not applicable to the WIP310, SPA301, or SPA501. Screen saver settings are found in the User tab on the Cisco SPA525G/525G2.
Screen Saver Wait
Amount of idle time before screen saver displays. NOTE Not applicable to the WIP310, SPA301, or SPA501. Screen saver settings are found in the User tab on the Cisco SPA525G/525G2.
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B
Description In screen saver mode, the phone LCD can display:
Parameter Screen Saver Icon
A background picture. The station time in the middle of the screen. A moving padlock icon. When the phone is locked, the status line displays a scrolling message "Press any key to unlock your phone." A moving phone icon. The station date and time in the middle of the screen.
NOTE Not applicable to the WIP310, SPA301, or SPA501. Screen saver settings are found in the User tab on the Cisco SPA525G/525G2. JPEG Logo Download URL (Cisco SPA525G/525G2) JPEG Wallpaper Download URL (Cisco SPA525G/525G2) Enable SMS URL from which to download a .jpg file for the phone logo display. URL from which to download a .jpg file for the phone wallpaper. Enables sending and receiving of SMS text messages on the phone. NOTE WIP310 only.
Line Key
When used in the configuration profile, parameters in this section must be appended with n, where n represents line 1, 2, 3, 4, 5 or 6. For more information on these parameters, see the Configuring Lines and Extensions section on page 33.
NOTE Does not apply to the WIP310.
Parameter Extension Short Name
Description Extension number of the line key. A short label shown on the LCD display for line key 1 through line key 6.
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B
Description Yes indicates that Line Key 1/2/3/4/5/6 is a shared call appearance. Otherwise this call appearance is not shared (it is private). Defaults to no.
Parameter Share Call Appearance
Extended Function
Use to assign Busy Lamp Field, Call Pickup, and Speed Dial Functions to Idle Lines on the IP phone. Syntax is:
fnc=type;sub=stationname@$PROXY;ext= extension#@$PROXY
where:
fnc: function blf: busy lamp field cp: call pickup sub: station name (not needed for speed dial) ext or usr: extension or user (the usr and ext keywords are interchangeable)
NOTE Not applicable to the WIP 310, SPA301, or SPA501.
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Miscellaneous Line Key Settings

Parameter SCA Line ID Mapping
Description Specifies the shared call appearance line ID mapping. Choose Vertical First or Horizontal First. Each LED can hold multiple calls and the first call on an LED makes it light up. Horizontal first means the second call makes the same LED flash. Vertical first means the second call lights up the next LED. For example, if Extension 101 is assigned to two LEDs, and Vertical First is selected, the second call on Extension 101 lights up the second LED. The third call makes the first LED flash, and the fourth call makes the second LED flash. If Horizontal First is selected, the second call on Extension 101 makes the first LED flash. The third call lights up the second LED, and the fourth call makes the second LED flash.
SCA Barge-In Enable
Enables the SCA Barge-In. Defaults to no.
SCA Sticky Auto Line Seize
When enabled, taking the phone off-hook will not automatically pick up an incoming call on a shared line.
Line Key LED Pattern

Parameter Idle LED
Description LED pattern during the Idle state, where the call appearance is not is in use and is available to make a new call. Leaving this entry blank indicates the default value of c=g.
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B
Description LED pattern during the Remote Undefined state, where the shared call state is undefined (the station is still waiting for the state information from the application server). Not applicable if the call appearance is not shared. Leaving this entry blank indicates the default value of c=r;p=d. LED pattern during the Local Seized state, where this station has seized the call appearance to prepare for a new outbound call. Leaving this entry blank indicates the default value of c=r. LED pattern during the Remote Seized state, where the shared call appearance is seized by another station. Not application if the call appearance is not shared. Leaving this entry blank indicates the default value of c=r;p=d. LED pattern during the Local Progressing state, where this station is attempting on this call appearance an outgoing call that is in proceeding (i.e. the called number is ringing). Leaving this entry blank indicates the default value of c=r. LED pattern during the Remote Progressing state, where another station is attempting on this shared call appearance an outbound call that is progressing. Not applicable if the call appearance is not shared. Leaving this entry blank indicates the default value of c=r;p=d. LED pattern during the Local Ringing state, when the call appearance is ringing. Leaving this entry blank indicates the default value of c=r;p=f. LED pattern during the Remote Ringing state, where the shared call appearance is in ringing on another station. Not applicable if the call appearance is not shared. Leaving this entry blank indicates the default value of c=r;p=d. LED pattern during the Local Active state, where the call appearance is engaged in an active call. Leaving this entry blank indicates the default value of c=r.
Parameter Remote Undefined LED
Local Seized LED
Remote Seized LED
Local Progressing LED
Remote Progressing LED
Local Ringing LED
Remote Ringing LED
Local Active LED
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B
Description LED pattern during the Remote Active state, where another station is engaged in an active call on this shared call appearance. Not applicable is this call appearance is not shared. Leaving this entry blank indicates the default value of c=r;p=d. LED pattern during the Local Held state, where the call appearance is held by this station. Leaving this entry blank indicates the default value of c=r;p=s. LED pattern during the Remote Held state, where another station has placed this call appearance on hold. Not applicable if the call appearance is not shared. Leaving this entry blank indicates the default value of c=r,p=s. LED pattern when the corresponding extension has failed to register with the proxy server. Leaving this entry blank indicates the default value of c=a. LED pattern when the Call Appearance is disabled (not available for any incoming or outgoing call). Leaving this entry blank indicates the default value of c=o. LED Pattern when the corresponding extension is trying to register with the proxy server. Leaving this entry blank indicates the default value of c=r;p=s. Call Back operation is currently active on this call appearance is not shared. Leaving this entry blank indicates the default value of c=r;p=s. LED pattern that indicates that a shared trunk is in use. LED pattern that indicates that a shared trunk is not in service. LED pattern to indicate that a shared trunk has been reserved.
Parameter Remote Active LED
Local Held LED
Remote Held LED
Register Failed LED
Disabled LED
Registering LED
Call Back Active LED
Trunk In-Use LED Trunk No Service LED
Trunk Reserved LED
Supplementary Services
Enable or disable the corresponding supplementary services on the phone. A value of yes indicates enabled; no indicates disabled.
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B
Description Enable/disable Three way conference service. Defaults to yes.
Parameter Conference Serv
Attn Transfer Serv
Enable/disable attended-call-transfer service. Defaults to yes.
Blind Transfer Serv
Enable/disable blind-call-transfer service. Defaults to yes.
DND Serv
Enable/disable do-not-disturb service. Defaults to yes.
Block ANC Serv
Enable/disable block-anonymous-call service. Defaults to yes.
Call Back Serv
Enable/disable call-back (a.k.a. repeating dialing) service. Defaults to yes.
Block CID Serv
Enable/disable blocking outbound Caller-ID service. Defaults to yes.
Secure Call Serv
Enable/disable secure-call service. Defaults to yes.
Cfwd All Serv
Enable/disable call-forward-all service. Defaults to yes.
Cfwd Busy Serv
Enable/disable call-forward-on-busy service. Defaults to yes.
Cfwd On No Ans Serv
Enable/disable call-forward-on-no-answer service. Defaults to yes.
Paging Serv
Enable/disable the paging service. Defaults to yes.
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B
Description Enable/disable the call park service. Defaults to yes.
Parameter Call Park Serv
Call Pick Up Serv
Enable/disable the call pickup service. Defaults to yes.
ACD Login Serv
Enable/disable the ACD Login Service, used for call centers. Typically enabled with the <SIP-B> parameter. Defaults to no.
Group Call Pick Up Serv
Enable/disable the group call pickup service. Defaults to yes.
Group Call Pick Up Serv
Enable/disable the group call pickup service. Defaults to yes.
ACD Ext
The extension used for handling ACD calls. Select from 1, 2, 3, 4, 5, or 6. Defaults to 1.
Service Annc Serv
Enable/disable sending announcement requests to a customer-supplied announcement server. Defaults to no.
Web Serv (Cisco SPA525G/ 525G2 only) SMS Serv (Cisco SPA525G/ 525G2 only)
Enable/disable the web server. Defaults to yes. Enable/disable the SMS text messaging server.
Ring Tone (Cisco SPA300 Series and Cisco SPA500 Series)

Each entry defines a ring tone to be used on the phone, with an ID between 1 and 10. The ID can be used in a DirEntry to indicate which ring tone to use when the corresponding caller calls.
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B
Description Ring tone script for ring 1. Defaults to n=Classic1;w=3;c=1. Ring tone script for ring 2. Defaults to n=Classic2;w=3;c=2. Ring tone script for ring 3. Defaults to n=Classic3;w=3;c=3. Ring tone script for ring 4. Defaults to n=Classic4;w=3;c=4. Ring tone script for ring 5. Defaults to n=Simple1;w=2;c=1. Ring tone script for ring 6. Defaults to n=Simple2;w=2;c=2. Ring tone script for ring 7. Defaults to n=Simple3;w=2;c=3. Ring tone script for ring 8. Defaults to n=Simple4;w=2;c=4. Ring tone script for ring 9. Defaults to n=Simple5;w=2;c=5. Ring tone script for ring 10. Defaults to n=Office;w=4;c=1.
Parameter Ring1
Ring2
Ring3
Ring4
Ring5
Ring6
Ring7
Ring8
Ring9
Ring10
Ring Tone (WIP310)
Parameter
Keypad Tone
Description
Select yes to enable the keypad tone to be played when a key on the keypad is pressed. Select no to silence the keypad. Corresponds to the volume of the keypad tone. Default is 5.
Keypad Tone Volume
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Auto Input Gain (dB)

Parameter Handset Input Gain
Description The amount of amplification to apply to the audio input signal for the handset. Defaults to zero.
Headset Input Gain
The amount of amplification to apply to the audio input signal for the headset. Defaults to zero. NOTE Not applicable to the SPA301 or the SPA501.
Speakerphone Input Gain
The amount of amplification to apply to the audio input signal for the speakerphone. Defaults to zero. NOTE Not applicable to the SPA301 or the SPA501.
Bluetooth Input Gain (SPA525G/525G2 only)
The amount of amplification to apply to the audio input signal for the Bluetooth device. Defaults to zero.
Handset Additional Input Gain
Applies additional input gain to the handset. NOTE Does not apply to the SPA525G/525G2.
Headset Additional Input Gain
Applies additional input gain to the headset. NOTE Does not apply to the SPA525G/525G2, SPA301, or the SPA501.
Speakerphone Additional Input Gain
Applies additional input gain to the speakerphone. NOTE Does not apply to the SPA525G/525G2, SPA301, or the SPA501.
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Multiple Paging Group Parameters
Parameter Group Paging Script
Description You can configure a phone as part of a paging group. Users can then direct pages to specific groups of phones. A phone can be part of no more than two paging groups, and user can page a maximum of five paging groups. The syntax is as follows: pggrp=ip-address:port;[name=xxx;]num=xxx; [listen={yes|no}]]; Where:
IP address: Multicast IP address of the phone that will

listen for and receive pages.
port: Port on which to page; you must use different

ports for each paging group. name (optional): The name of the paging group. num: The number users will dial to access the paging group; must be unique to the group. listen: If the phone should listen on the page group. Only the first two groups with listen set to yes will listen to group pages. If the field is not defined, the default value is no, so you must set this field to listen to the group pages.
BroadSoft Settings
The Cisco SPA300 Series and Cisco SPA500 Series supports the BroadSoft directory feature and synchronization of Do Not Disturb and Call Forward. The following configuration fields are available:
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B
Description Set to yes to enable BroadSoft directory for the phone user. Defaults to no. NOTE Not applicable to the Cisco SPA301 or Cisco SPA501.
Parameter Directory Enable
XSI Host Server
Enter the name of the server; for example, xsp.xdp.broadsoft.com. NOTE Not applicable to the Cisco SPA301 or Cisco SPA501.
Directory Name
Name of the directory. Displays on the users phone as a directory choice. NOTE Not applicable to the Cisco SPA301 or Cisco SPA501.
Directory Type
Select the type of BroadSoft directory:
Enterprise (default): Allows users to search on last name, first name, user or group ID, phone number, extension, department, or email address. Group: Allows users to search on last name, first name, user ID, phone number, extension, department, or email address. Personal: Allows users to search on last name, first name, or telephone number.
NOTE Not applicable to the Cisco SPA301 or Cisco SPA501. Directory UserID BroadSoft User ID of the phone user; for example, [email protected]. NOTE Not applicable to the Cisco SPA301 or Cisco SPA501. Directory Password Alphanumeric password associated with the User ID. NOTE Not applicable to the Cisco SPA301 or Cisco SPA501.
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B
Description Allows the phone to synchronize with the call server so that if Do Not Disturb or Call Forwarding settings are changed on the phone, changes are also made on the server; if changes are made on the server, they are propagated to the phone. This feature is disabled by default. Choose the extension (1 through 5) that is registered to the BroadSoft server.
Parameter Call Feature Sync Ext
Lightweight Directory Access Protocol (LDAP) Corporate Directory Search

NOTE Does not apply to the WIP310, SPA301, or SPA501.
If using Active Directory with authentication set to MD5, you must first configure the following: Click the System tab. In the Optional Network Configuration section, under Primary DNS, enter the IP address of the DNS server. In the Optional Network Configuration section, under Domain, enter the LDAP domain.
Parameter LDAP Dir Enable LDAP Corp Dir Name
Description Choose yes to enable LDAP. Enter a free-form text name, such as Corporate Directory. Enter a fully qualified domain name or IP address of LDAP server, in the following format:
nnn.nnn.nnn.nnn
LDAP Server
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B
Description Select the authentication method that the LDAP server requires. Choices are:
Parameter LDAP Auth Method
NoneNo authentication is used between the client and the server. SimpleThe client sends its fully-qualified domain name and password to the LDAP server. May present security issues. Digest-MD5The LDAP server sends authentication options and a token to the client. The client returns an encrypted response that is decrypted and verified by the server.
LDAP Client DN
Enter the distinguished name domain components [dc] ; for example:

dc=cv2bu,dc=com
If using the default Active Directory schema (Name(cn)->Users->Domain), an example of the client DN follows:
cn="David Lee",dc=users,dc=cv2bu,dc=com
LDAP Username
Enter the username for a credentialed user on the LDAP server. Enter the password for the LDAP username. Specify a starting point in the directory tree from which to search. Separate domain components [dc] with a comma. For example: dc=cv2bu,dc=com
LDAP Password LDAP Search Base
LDAP Last Name Filter
This defines the search for surnames [sn], known as last name in some parts of the world. For example, sn:(sn=*$VALUE*). This search allows the provided text to appear anywhere in a name, beginning, middle, or end. This defines the search for the common name [cn]. For example, cn:(cn=*$VALUE*). This search allows the provided text to appear anywhere in a name, beginning, middle, or end.
LDAP First Name Filter
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B
Description Additional customized search item. Can be blank if not needed. Customized filter for the searched item. Can be blank if not needed. Additional customized search item. Can be blank if not needed. Customized filter for the searched item. Can be blank if not needed. Format of LDAP results display on phone where:
Parameter LDAP Search Item 3
LDAP Item 3 Filter
LDAP Search Item 4
LDAP Item 4 Filter
LDAP Display Attrs
aAttribute name cnCommon name snSurname (last name) telephoneNumberPhone number nDisplay name
For example, n=Phone will cause "Phone:" to be displayed in front of the phone number of an LDAP query result when the detail soft button is pressed. ttype
When t=p, that is, t is of type phone number, then the retrieved number can be dialed. Only one number can be made dialable. If two numbers are defined as dialable, only the first number is used. For example, a=ipPhone, t=p; a=mobile, t=p; This example results in only the IP Phone number being dialable and the mobile number will be ignored.
pphone number
When p is assigned to a type attribute, example t=p, then the retrieved number will be dialable by the phone.
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B
Description Can be blank if not needed. NOTE With the LDAP number mapping you can manipulate the number that was retrieved from the LDAP server. For example, you can append 9 to the number if your dial plan requires a user to enter 9 before dialing. Add the 9 prefix by adding (<:9xx.>) to the LDAP Number Mapping field. For example, 555 1212 would become 9555 1212. If you do not manipulate the number in this fashion, a user can use the Edit Dial feature to edit the number before dialing out.
Parameter LDAP Number Mapping
XML Service
The Cisco SPA300 Series and the Cisco SPA500 Series IP Phones support XML services, such as an XML Directory Service or other XML applications. (Not applicable to the Cisco SPA301 or the Cisco SPA501.) The following configuration fields are available:
Parameter XML Directory Service Name
Description Name of the XML Directory. Displays on the users phone as a directory choice. URL where the XML Directory is located. Name of the XML application. Displays on the users phone as a web application choice. URL where the XML application is located.
XML Directory Service URL XML Application Service
XML Application Service URL
Extension Mobility
For more information, see Configuring Extension Mobility, page 94.
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B
Description Enable or disable extension mobility. Defaults to no (disabled).
Parameter Extension Mobility
EM User Domain
The user domain for extension mobility. Defaults to blank.
Programmable Softkeys
The Cisco SPA300 Series and Cisco SPA500 Series IP phones (models with display screens) have four softkeys on the screen that, when pressed, perform certain actions. You can customize the softkeys displayed on the phone, and create your own softkeys for speed dials or XML scripts. Customized softkey information is entered in the PSK1 through PSK6 fields.
Parameter Programmable Softkey Enable
Description Enables programmable softkeys (SPA525G/525G2 only). Softkeys that display when the phone is idle. See Customizing Phone Softkeys, page 62 for more information. Softkeys that display when a call has been missed. See Customizing Phone Softkeys, page 62 for more information. Softkeys that display when the receiver is lifted, or the headphone or speakerphone buttons are pressed. See Customizing Phone Softkeys, page 62 for more information. Softkeys that display when the user must enter dialing data. See Customizing Phone Softkeys, page 62 for more information.
Idle Key List
Missed Call Key List
Off Hook Key List
Dialing Input Key List
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B
Description Softkeys that display when a call is attempting to connect. See Customizing Phone Softkeys, page 62 for more information. (SPA525G/525G2 only)
Parameter Progressing Key List
Connected Key List
Softkeys that display when a call is connected. See Customizing Phone Softkeys, page 62 for more information. Softkeys that display when a call transfer has been initiated. See Customizing Phone Softkeys, page 62 for more information. Softkeys that display when a conference call has been initiated. See Customizing Phone Softkeys, page 62 for more information. Softkeys that display when a conference call is in progress. See Customizing Phone Softkeys, page 62 for more information. Softkeys that display when a call is disconnecting. See Customizing Phone Softkeys, page 62 for more information. (SPA525G/525G2 only)
Start-Xfer Key List
Start-Conf Key List
Conferencing Key List
Releasing Key List
Hold Key List
Softkeys that display when one or more calls are on hold. See Customizing Phone Softkeys, page 62 for more information. (SPA525G/525G2 only)
Ringing Key List
Softkeys that display when a call is incoming. See Customizing Phone Softkeys, page 62 for more information. Softkeys that display when a call is active on a shared line. See Customizing Phone Softkeys, page 62 for more information. (SPA525G/525G2 only)
Shared Active Key List
Shared Held Key List
Softkeys that display when a call is on hold on a shared line. See Customizing Phone Softkeys, page 62 for more information. (SPA525G/525G2 only)
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B
Description To configure a speed dial script, enter the following in the PSK field: fnc=sd;ext=extensionname@$PROXY;vid=outbound extnum;nme=name where fnc is the function of the key (speed dial), ext (extensionname) is the extension being dialed, vid is the extension on the calling phone from which the outbound call is sent, and name is the name of the speed dial being configured. To configure an XML script, enter the following in the PSK field: fnc=xml;url=http://scriptURL.xml;nme=scriptname where fnc is the function of the key (an XML script), scriptURL.xml is the URL where the script is located, and scriptname is the name of the script.
Parameter PSK1 through PSK6
Ext Tab
The Ext tabs vary by phone and depend on the number of extensions the phone model supports. This section describes the fields for the following Ext tab headings: General, page 274 Share Line Appearance, page 295 NAT Settings, page 296 Network Settings, page 296 SIP Settings, page 297 Call Feature Settings, page 301 Proxy and Registration, page 304 Subscriber Information, page 306 Audio Configuration, page 307 Dial Plan, page 310
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In a configuration profile, the Line parameters must be appended with the appropriate numeral to indicate the line to which the setting applies. For example:
[1] to specify line one [2] to specify line two
General
Line Enable: To enable this line for service, select yes. Otherwise, select no. Defaults to yes.
Share Line Appearance
Parameter Share Ext
Description Indicates whether this extension is to be shared with other stations or private. If the extension is not shared, then a call appearance assigned to this extension is not shared, regardless the setting of <Share Call Appearance> for that call appearance. If the extension is shared, then whether or not a call appearance assigned to this extension is shared follows the setting of <Share Call Appearance> for that call appearance. The choices are shared or private. Defaults to shared.
Shared User ID
The user identified assigned to the shared line appearance. Number of seconds before the SIP subscription expires. Before the subscription expiration, the phone gets NOTIFY messages from the SIP server on the status of the shared phone extension. Defaults to 60 seconds. When enabled, the message waiting indicator lights only for messages on private lines. This field is for future use.
Subscription Expires
Restrict MWI
Monitor User ID (Cisco SPA300 Series, Cisco SPA500 Series)
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NAT Settings
Parameter NAT Mapping Enable
Description To use externally mapped IP addresses and SIP/RTP ports in SIP messages, select yes. Otherwise, select no. Defaults to no.
NAT Keep Alive Enable
To send the configured NAT keep alive message periodically, select yes. Otherwise, select no. Defaults to no.
NAT Keep Alive Msg
Enter the keep alive message that should be sent periodically to maintain the current NAT mapping. If the value is $NOTIFY, a NOTIFY message is sent. If the value is $REGISTER, a REGISTER message without contact is sent. Defaults to $NOTIFY.
NAT Keep Alive Dest
Destination that should receive NAT keep alive messages. If the value is $PROXY, the messages are sent to the current or outbound proxy. Defaults to $PROXY.
Network Settings
Parameter SIP TOS/DiffServ Value
Description TOS/DiffServ field value in UDP IP packets carrying a SIP message. Defaults to 0x68.
SIP CoS Value
CoS value for SIP messages. Defaults to 3.
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B
Description ToS/DiffServ field value in UDP IP packets carrying RTP data. Defaults to 0xb8.
Parameter RTP TOS/DiffServ Value
RTP CoS Value
CoS value for RTP data. Defaults to 6.
Network Jitter Level
Determines how jitter buffer size is adjusted by the IP phone. 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. Defaults to high.
Jitter Buffer Adjustment
Controls how the jitter buffer should be adjusted. Select the appropriate setting: up and down, up only, down only, or disable. Defaults to up and down.
SIP Settings
Parameter SIP Transport
Description Select from UDP, TCP, or TLS. Defaults to UDP.
SIP Port
Port number of the SIP message listening and transmission port. Defaults to 5060.
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B
Description 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. Defaults to no.
Parameter SIP 100REL Enable
EXT SIP Port Auth Resync-Reboot
The external SIP port number. If this feature is enabled, the IP phone authenticates the sender when it receives the NOTIFY resync reboot (RFC 2617) message. To use this feature, select yes. Otherwise, select no. Defaults to yes.
SIP Proxy-Require
The SIP proxy can support a specific extension or behavior when it sees this header from the user agent. If this field is configured and the proxy does not support it, it responds with the message, unsupported. Enter the appropriate header in the field provided. To use the Remote-Party-ID header instead of the From header, select yes. Otherwise, select no. Defaults to yes.
SIP Remote-Party-ID
Referor Bye Delay
Controls when the IP phone 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. Defaults to 4.
Refer-To Target Contact
To contact the refer-to target, select yes. Otherwise, select no. Default: no
Referee Bye Delay
For the Referee Bye Delay, enter the appropriate period of time in seconds. Defaults to 0.
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B
Description SIP messages are received at or sent from the proxy listen port. This feature controls which SIP messages to log. Choices are as follows:
Parameter SIP Debug Option
noneNo logging. 1-lineLogs the start-line only for all messages. 1-line excl. OPTLogs the start-line only for all messages except OPTIONS requests/responses. 1-line excl. NTFYLogs the start-line only for all messages except NOTIFY requests/responses. 1-line excl. REGLogs the start-line only for all messages except REGISTER requests/responses. 1-line excl. OPT|NTFY|REGLogs the start-line only for all messages except OPTIONS, NOTIFY, and REGISTER requests/responses. fullLogs all SIP messages in full text. full excl. OPTLogs all SIP messages in full text except OPTIONS requests/responses. full excl. NTFYLogs all SIP messages in full text except NOTIFY requests/responses. full excl. REGLogs all SIP messages in full text except REGISTER requests/responses. full excl. OPT|NTFY|REGLogs all SIP messages in full text except for OPTIONS, NOTIFY, and REGISTER requests/responses.
Defaults to none. Refer Target Bye Delay For the Refer Target Bye Delay, enter the appropriate period of time in seconds. Defaults to 0. Sticky 183 If this feature is enabled, the IP telephony ignores further 180 SIP responses after receiving the first 183 SIP response for an outbound INVITE. To enable this feature, select yes. Otherwise, select no. Defaults to no.
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B
Description When enabled, authorization is required for initial incoming INVITE requests from the SIP proxy. If set to yes, as a transferee, the phone will send a NOTIFY with Event:Refer to the transferor for any 1xx response returned by the transfer target, on the transfer call leg. If set to no, the phone will only send a NOTIFY for final responses (200 and higher). NOTE Not applicable to the WIP310.
Parameter Auth INVITE
Ntfy Refer On 1xx-To-Inv
Use Anonymous With RPID
This parameter applies only if <SIP Remote-Party-ID> is set to yes; otherwise, it is ignored. If the parameter is set to yes, the FROM header's display-name and user-id fields are set to anonymous when the caller blocks his caller-id. If the parameter is set to no, the FROM header's display-name and user-id are not masked. The Remote-Party-ID header indicates privacy=full when the caller wishes to block his callerid. Default: yes. NOTE Not applicable to the WIP310.
Set G729annexb
Configure G.729 Annex B settings. NOTE Not applicable to Cisco SPA525G.
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Ext Tab
B
Description For configuration of a SIP event package, SIP PUBLISH, that enables the collection and reporting of metrics that measure the quality for VoIP sessions. Voice call quality information derived from RTCP-XR and call information from SIP is conveyed from a User Agent in a session to the third party in SIP PUBLISH method. To configure, first configure RTCP-XR (see the RTP Parameters section on page 247). For example, you need to configure the RTCP Tx Interval. In the Voice Quality Report Address field, enter the name of the collector that will collect the statistics from the SIP PUBLISH events. For example, enter collector@fullyqualified-domain-name ([email protected]) or collector@IP-address ([email protected]). After RTCP-XR feature is enabled, the call status information is updated on the Voice > Info page during an active call. Additionally, RTCP-XR packets containing a voice metrics block report will be sent with the interval specified in the RTCP Tx Interval. When the call session is ended, a SIP PUBLISH with voice metrics info is sent to the collector endpoint.
Parameter Voice Quality Report Address
Call Feature Settings
Parameter Blind Attn-Xfer Enable
Description Enables the IP phone 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 IP phone 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. Defaults to no.
MOH Server
User ID or URL of the auto-answering streaming audio server. When only a user ID is specified, the current or outbound proxy is contacted. Music-on-hold is disabled if the MOH Server is not specified. Defaults to imusic when used with a Cisco SPA 9000 IP PBX.
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Ext Tab
B
Description Indicates whether the Message Waiting Indicator on the phone is lit. This parameter is toggled by a message from the SIP proxy to indicate if a message is waiting. You can manually modify it to clear or set the flag in the Ext 1-6 tab. 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. Defaults to No.
Parameter Message Waiting
Auth Page
Specifies whether to authenticate the invite before auto answering a page. Defaults to No.
Default Ring
Type of ring heard. This corresponds to the Ring Tone on the Phone tab. Choose from No Ring, 1 through 10, User 1, or User 2. Defaults to 1.
Auth Page Realm
Identifies the Realm part of the Auth that is accepted when the Auth Page parameter is set to Yes. This parameter accepts alphanumeric characters. Defaults to blank.
Conference Bridge URL
This is the URL used to join into a conference call, generally in the form of the word conference or user@IPaddress:port. Defaults to blank.
Auth Page Password
Identifies the password used when the Auth Page parameter is set to Yes. This parameter accepts alphanumeric characters. Defaults to blank.
Mailbox ID
Identifies the voice mailbox number/ID for the phone. Defaults to blank.
Voice Mail Server
Identifies the SpecVM server for the phone, generally the IP address and port number of the VM server. The expiration time, in seconds, of a subscription to a voice mail server.
Voice Mail Subscribe Interval
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Ext Tab
B
Description Reserved feature. Specifies whether to enable a SIP-B feature regarding the sending of a Notify to the phone when a call is forwarded elsewhere. Defaults to No.
Parameter State Agent CFWD Notify Serv
CFWD Notifier
Typically, this field is configured with the SIP proxy information.
User ID with Domain (Cisco SPA300 Series)
When this field is set to yes, the IP phone will show the Caller ID followed by domain name, and the domain name is also shown in the received calls list. This parameter is used when calls are made between different branches of the same phone system. For example, if user [email protected] receives a call from [email protected], by default the phone will only show the call as being from Mary, so John is not able to pick up or call back Mary from the received call list. With this parameter set to yes, the phone logs the call as being from [email protected], and John can dial Mary from the received call list.
Proxy and Registration
Parameter Proxy
Description SIP proxy server and port number set by the Service Provider for all outbound requests. For example: 192.168.2.100:6060. SIP Outbound Proxy Server where all outbound requests are sent as the first hop.
Outbound Proxy
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Ext Tab
B
Description Enables an outbound proxy (for example, 172.20.2.1:5060port is optional) or a domain name such as sip.server.com as long as this name is a fullyqualified domain name. If set to no, the Outbound Proxy and Use OB Proxy in Dialog fields are ignored. Defaults to no. Optionally, the proxy can be configured (SPA5xx only) for Survivable Remote Site Telephony (SRST) support. The proxy is configured with an extension that includes a statically-configured DNS SRV record or DNS A record. Configuring the proxy allows for failover and fallback functionality with a secondary proxy server. For example: For SRV Record: sip.server.com:SRV=node1.sip.server.com:5060:p=1:w= 50|node2.sip.server.com:5060:p=2:w=50 NOTE Set Use DNS SRV to no and DNS SRV Auto Prefix to no. For A Record: sip.server.com:A=172.20.2.1,172.20.2.2 NOTE Set Use DNS SRV to no and DNS SRV Auto Prefix to no.
Parameter Use Outbound Proxy
Use OB Proxy In Dialog
Whether to force SIP requests to be sent to the outbound proxy within a dialog. Ignored if <Use Outbound Proxy> is no or <Outbound Proxy> is empty. Defaults to yes.
Register
Enable periodic registration with the <Proxy>. This parameter is ignored if <Proxy> is not specified. Defaults to yes.
Make Call Without Reg
Allow making outbound calls without successful (dynamic) registration by the unit. If no, the dial tone will not play unless registration is successful. Defaults to no.
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Ext Tab
B
Description Allow answering inbound calls without successful (dynamic) registration by the unit. If proxy responded to REGISTER with a smaller Expires value, the phone will renew registration based on this smaller value instead of the configured value. If registration failed with an Expires too brief error response, the phone will retry with the value given in the Min-Expires header in the error response. Defaults to 60.
Parameter Register Expires
Ans Call Without Reg
If enabled, the user does not have to be registered with the proxy to answer calls. Defaults to no.
Use DNS SRV
Whether to use DNS SRV lookup for Proxy and Outbound Proxy. Defaults to no.
DNS SRV Auto Prefix
If enabled, the phone will automatically prepend the Proxy or Outbound Proxy name with _sip._udp when performing a DNS SRV lookup on that name. Defaults to no.
Proxy Fallback Intvl
This parameter sets the delay (sec) after which the phone will retry from the highest priority proxy (or outbound proxy) servers after it has failed over to a lower priority server. This parameter is useful only if the primary and backup proxy server list is provided to the phone via DNS SRV record lookup on the server name. (Using multiple DNS A record per server name does not allow the notion of priority and so all hosts will be considered at the same priority and the phone will not attempt to fall back after a fail over). Defaults to 3600
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Ext Tab
B
Description Select Normal or Based on SRV port. The phone creates an internal list of proxies returned in the DNS SRV records. If you select Normal, the list contains proxies ranked by weight and priority. If you select Based on SRV, the phone uses normal, then inspects the port number based on the first listed proxy port. Defaults to Normal.
Parameter Proxy Redundancy Method
Subscriber Information
Parameter Display Name User ID Password
Description Display name for caller ID. Extension number for this line. Password for this line. Defaults to blank.
Use Auth ID
To use the authentication ID and password for SIP authentication, select yes. Otherwise, select no to use the user ID and password. Defaults to no.
Auth ID
Authentication ID for SIP authentication. Defaults to blank.
Mini Certificate
Base64 encoded of Mini-Certificate concatenated with the 1024-bit public key of the CA signing the MC of all subscribers in the group. Defaults to blank.
SRTP Private Key
Base64 encoded of the 512-bit private key per subscriber for establishment of a secure call. Defaults to blank.
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Ext Tab
Audio Configuration
A codec resource is considered as allocated if it has been included in the SDP codec list of an active call, even though it eventually may not be the one chosen for the connection. So, if the G.729a codec is enabled and included in the codec list, that resource is tied up until the end of the call whether or not the call actually uses G.729a. If the G.729a resource is already allocated and since only one G.729a resource is allowed per device, no other low-bit-rate codec may be allocated for subsequent calls; the only choices are G711a and G711u. On the other hand, two G.723.1/G.726 resources are available per device. Therefore it is important to disable the use of G.729a in order to guarantee the support of two simultaneous uses of the G.723/G.726 codecs.
Parameter Preferred Codec
Description Preferred codec for all calls. (The actual codec used in a call still depends on the outcome of the codec negotiation protocol.) Select one of the following: G711u, G711a, G722, G726-16, G726-24, G726-32, G726-40, G729a, or G723. NOTE Cisco SPA525G/525G2 choices are: G711u, G711a, G726-32, G729a, and G722. G.723 not available on SPA300 Series or SPA500 Series. G722 not available on WIP310. Defaults to G711u.
Use Pref Codec Only
To use only the preferred codec for all calls, select yes. (The call fails if the far end does not support this codec.) Otherwise, select no. Defaults to no.
Second Preferred Codec
The second preferred codec when the preferred codec cannot be used. If Use Pref Codec Only is enabled (set to yes), this parameter is not used. Defaults to Unspecified.
Third Preferred Codec
The third preferred codec when the preferred codec and second preferred codec cannot be used. If Use Pref Codec Only is enabled (set to yes), this parameter is not used. Defaults to Unspecified.
307

Ext Tab
B
Description To enable the use of the G.729a codec at 8 kbps, select yes. Otherwise, select no. Defaults to yes.
Parameter G729a Enable
G722 Enable
Enables use of the G.722 codec. Defaults to yes. NOTE Not applicable to the WIP310.
G723 Enable
To enable the use of the G.723a codec at 6.3 kbps, select yes. Otherwise, select no. Defaults to yes. NOTE G.723 is not supported on the Cisco SPA300 Series, Cisco SPA500 Series, or WIP310.
G726-16 Enable
To enable the use of the G.726 codec at 16 kbps, select yes. Otherwise, select no. Defaults to yes. NOTE Not supported on the Cisco SPA525G/525G2.
G726-24 Enable
To enable the use of the G.726 codec at 24 kbps, select yes. Otherwise, select no. Defaults to yes. NOTE Not supported on the Cisco SPA525G/525G2 or WIP310.
L 16 Enable
To enable the use of the L16 codec, select yes. Otherwise, select no. Defaults to yes. NOTE Cisco SPA525G/525G2 only.
G726-32 Enable
To enable the use of the G.726 codec at 32 kbps, select yes. Otherwise, select no. Defaults to yes.
G726-40 Enable
To enable the use of the G.726 codec at 40 kbps, select yes. Otherwise, select no. Defaults to yes. NOTE Not applicable to the Cisco SPA525G/525G2.
308

Ext Tab
B
Description Allows the release of codecs not used after codec negotiation on the first call so that other codecs can be used for the second line. To use this feature, select yes. Defaults to yes.
Parameter Release Unused Codec
DTMF Process AVT
Select yes to process RTP DTMF events. Otherwise, select no. If this parameter is set to no, the AVT payload type is not included in outbound SDP. Defaults to yes.
Silence Supp Enable
To enable silence suppression so that silent audio frames are not transmitted, select yes. Otherwise, select no. See Ensuring Voice Quality, page 132. Defaults to no.
DTMF Tx Method
Select the method to transmit DTMF signals to the far end: InBand, AVT, INFO, Auto, InBand+INFO, or AVT+INFO. InBand sends DTMF using the audio path. AVT sends DTMF as AVT events. INFO uses the SIP INFO method. Auto uses InBand or AVT based on the outcome of codec negotiation. Defaults to Auto.
DTMF Tx Volume for AVT Packet
Allows you to manually configure the AVT Tx volume. The value of this parameter is inserted into the volume field of the payload in the AVT packet. Values are based on the AVT specification as described in RFC 2833, RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals. According to RFC 2833, the volume field is represented by 6 bits, and describes the power level of the tone, expressed in dBm0 after dropping the sign. Valid range for this parameter is 0 to 63. If the provisioned value is negative, it will be negated first. Thereafter, if the value is beyond the high limit of 63, it will be clipped to 63. The default value is 0, and is the recommended setting. However, some gateways do not accept this volume setting. If the gateway does not accept the value of 0, the DTMF tone is not relayed to the remote end. As a workaround for the phone to interoperate with those gateways, you can change the value to a value greater than 0.
309

Ext Tab
B
Description If set to yes, the phone communicates using the remote phones preferred codec. If set to no, the Cisco IP phone communicates using its own preferred codec (as indicated in the Preferred Codec field and in the SDP by order of preferences). The default vale is no. When set to Default, the Cisco IP phone responds to an Invite with a 200 OK response advertising the preferred codec only. When set to List All, the Cisco IP phone responds listing all the codecs that the phone supports. The default value is Default, or to respond with the preferred codec only.
Parameter
Use Remote Pref Codec
Codec Negotiation
A codec resource is considered allocated if it has been included in the SDP codec list of an active call, even though it eventually might not be chosen for the connection. If the G.729a codec is enabled and included in the codec list, that resource is tied up until the end of the call whether or not the call actually uses G.729a. If the G729a resource is already allocated (and since only one G.729a resource is allowed per phone), no other low-bit-rate codec can be allocated for subsequent calls. The only choices are G711a and G711u. Since two G.723.1/G.726 resources are available per IP phone, you should disable the use of G.729a to guarantee support for two simultaneous G.723/G.726 codecs.
Dial Plan
The default dial plan script for each line is as follows: (*xx|[3469]11|0|00|[29]xxxxxx|1xxx[2-9]xxxxxx|x xxxxxxxxxxx.).
310

User Tab
B
Description Dial plan script for this line. The default is (<9:>xx.) (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[29]xxxxxxS0|xxxxxxxxxxxx.) The dial plan syntax is expanded in the Cisco SPA to allow the designation of three parameters to be used with a specific gateway:
Parameter Dial Plan
uid the authentication user-id pwd the authentication password nat if this parameter is present, use NAT mapping
Each parameter is separated by a semi-colon (;). Caller ID Map Inbound caller ID numbers can be mapped to a different string. For example, a number that begins with +44xxxxxx can be mapped to 0xxxxxx. This feature has the same syntax as the Dial Plan parameter. With this parameter, you can specify how to map a caller ID number for display on screen and recorded into call logs. (Not applicable to WIP310.) Enable or disable IP dialing. Defaults to no.
Enable IP Dialing
User Tab
This section describes the fields for the following headings on the User tab: Call Forward, page 312 Speed Dial, page 312 Supplementary Services, page 313 Camera Settings (Cisco SPA525G/525G2), page 313 Web Information Service Settings (Cisco SPA525G/525G2), page 313
311

User Tab
Audio Volume, page 313 Screen (Cisco SPA525G/525G2), page 314
Call Forward
Parameter Cfwd All Dest Cfwd Busy Dest
Description Enter the extensions to forward calls to. Enter the extensions to forward calls to when the line is busy. Defaults to voice mail.
Cfwd No Ans Dest
Enter the extension to forward calls to when the call is not answered. Defaults to voice mail.
Cfwd No Ans Delay
Enter the time delay in seconds to wait before forwarding a call that is not answered. Defaults to 20 seconds.
See Vertical Service Activation Codes, page 260 for more information on call forwarding parameters.
Speed Dial
Speed Dial 2 through 9: Target phone number (or URL) assigned to speed dial 2, 3, 4, 5, 6, 7, 8, or 9. Defaults to blank.
NOTE Speed dial configuration has its own tab on the Cisco SPA525G and does not
appear in this section on the WIP310. Speed dial configuration from the WIP310 is done on the phone. You can also configure speed dials on the Cisco SPA300 Series and Cisco SPA500 Series IP phones; see the User Guide for the phone for more information.
312

User Tab
Supplementary Services
The IP phone provides native support of a large set of enhanced or supplementary services. All of these services are optional. Most supplementary service parameters are listed in Supplementary Services, page 281. The user can enable or disable supplementary services and other settings in this section. A supplementary service should be disabled if the user has not subscribed for it, or the service provider intends to support similar service using other means. For more star code or supplementary service information, see Configuring Supplementary Services (Star Codes), page 181.
Camera Settings (Cisco SPA525G/525G2)

The Cisco SPA525G/525/G2 works with the Cisco WVC2300 Wireless-G Business Internet Video Camera and the Cisco PVC2300 Business Internet Video Camera to provide simple video monitoring from your IP phone. See Entering Camera Information Into the Cisco SPA525G/525G2 Configuration Utility, page 96.
Web Information Service Settings (Cisco SPA525G/525G2)

NOTE These parameters apply only to the Cisco SPA525G/525G2.
For configuration information, see Configuring RSS Newsfeeds on the Cisco SPA525G/525G2 IP Phone, page 74.
Audio Volume
Parameter
Ringer Volume Speaker Volume Handset Volume
Description
Sets the default volume for the ringer. Sets the default volume for the full-duplex speakerphone. Sets the default volume for the handset.
313

User Tab
B
Sets the default volume for the headset. Volume of the Bluetooth device. NOTE Applies to the Cisco SPA525G/525G2 only.
Headset Volume Bluetooth Volume
Screen (Cisco SPA525G/525G2)
Parameter
Screen Saver Enable
Description
Enables a screen saver on the phones LCD. When the phone is idle for a specified time, it enters screen saver mode. (Users can set up screen savers directly using phone Setup button.) Any button press or on/off hook event triggers the phone to return to its normal mode. (The screen shows Press any key to unlock your phone.) If a user password is set, the user must enter it to exit screen saver mode. Choose the type of screen saver:
Screen Saver Type

Screen Saver Trigger Time Screen Saver Refresh Time
Black BackgroundDisplays a black screen. Gray BackgroundDisplays a gray screen. Black/Gray RotationThe screen incrementally cycles from black to gray. Picture RotationThe screen rotates through available pictures on the phone. Digital FrameShows the background picture.
Number of seconds that the phone remains idle before the screen saver turns on. Number of seconds before the screen saver should refresh (if, for example, you chose a rotation of pictures).
314

User Tab
B
Description
Text logo to display when the phone boots up. A service provider, for example, can enter logo text as follows:
Parameter
Text Logo
Up to 2 lines of text Each line must be fewer than 32 characters Insert a new line character (\n) between lines Insert escape code %0a
For example, Super\n%0aTelecom will display:
Super Telecom
For more information, see the Configuring Phone Information and Display Settings section on page 46. BMP Picture Download URL URL locating the bitmap (.BMP) or .jpg file to display on the LCD background. For more information, see the Configuring Phone Information and Display Settings section on page 46. Logo Type Select from Default, Download BMP Picture, or Text Logo. Defaults to Default. For more information, see the Configuring Phone Information and Display Settings section on page 46. Background Picture Type Select from Default, Download BMP Picture, or None. Defaults to Default. For more information, see the Configuring Phone Information and Display Settings section on page 46. LCD Contrast Enter a number value from 1 to 30. The higher the number, the greater the contrast on the screen. Select yes to enable the screen back light. Enter the number of seconds before the back light should turn off.
Back Light Enable Back Light Timer (sec)
315


This tab includes the following sections: General, page 316 Unit 2, page 317
General
Parameter Subscribe Expires Description Specifies how long the subscription remains valid. After the specified period of time, elapses, the Cisco SPA500S initiates a new subscription. Defaults to 1800. Subscribe Retry Interval Specifies the length of time to wait to try again if subscription fails. Enables or disables the first Cisco SPA500S unit (each IP phone can have up to two Cisco SPA500Ss attached). Length of delay before attempting to subscribe. Defaults to 1. Unit 2 Enable Enables or disables the second Cisco SPA500S unit (each IP phone can have up to two Cisco SPA500Ss attached). Selects the type of server used (Cisco SPA 9000, BroadSoft, or Asterisk). Enables or disables test mode. When test mode is enabled, the LEDs are turned on when keys are pressed, going from off to green to red, and back to off. In test mode, when all the buttons on the Cisco SPA500S are returned to off, all the keys become orange. The IP phone must be rebooted after the test is completed. The star code used for picking up a ringing call. Defaults to *98.
Unit 1 Enable
Subscribe Delay
Server Type
Test Mode Enable
Attendant Console Call Pickup Code
316

Attendant Console Status
B
Description Automatically configures BLF subscriptions for all users on a monitored list. See Configuring BroadSoft Busy Lamp Field Auto-Configuration, page 217. Enter a strings that define the extension and other parameters associated with each lighted button on the first Cisco SPA500S unit. Keywords and values are case-sensitive. The configuration script is described in the Unit/Key Configuration Scripts section on page 214.
Parameter BLF List URI
Unit 1 Key 1-32
For more information, see Chapter 9, Configuring the Cisco SPA500S Attendant Console..
Unit 2
See the description for Unit 1 above. Enter a strings that define the extension and other parameters associated with each lighted button on the second Cisco SPA500S unit. Keywords and values are case-sensitive. The configuration script is described in the Setting Up the Cisco SPA500S Attendant Console section on page 210.
Attendant Console Status

This page provides two tabs to display the status of up to two Cisco SPA500S units that are supported by a single IP phone: Unit 1Displays information about the first Cisco SPA500S unit. Unit 2Displays information about the second Cisco SPA500S unit.
317

Cisco SPA525G/525G2-Specific Tabs
B
Description Displays if the Unit is enabled or disabled. Displays when the current subscription expires. After the subscription expires, the Cisco SPA500S automatically requests a new subscription. Displays the version of the hardware. Displays whether the unit is powered on and connected or not. Displays the length of time the Cisco SPA500S waits to try again if subscription fails. Displays the version of the software currently running on the unit. Displays the name assigned to each key (1-32) on the Cisco SPA500S unit. Displays the function enabled for each key (1-32) on the Cisco SPA500S unit. Displays the extension assigned to each key (1-32) on the Cisco SPA500S unit. Displays the subscribe URI configured for each key (132) on the Cisco SPA500S unit.
Each tab provides the read-only fields described in the following table:
Parameter Unit Enable Subscribe Expires
HW Version Unit Online
Subscribe Retry Interval
SW Version
Key Name
Type
Line
Station

The following tabs appear on the Cisco SPA525G/525G2.
Wi-Fi
Enable or disable the Wireless-G service on the phone from this tab.
318

B
Description
Click On to enable the wireless controller. Choose the method of wireless setup:
Parameter
Wireless Enable Wi-Fi Device
Wi-Fi ProfileCreate a wireless profile by manually entering information. Wi-Fi Protected SetupIf your router has a WPS button, you can use Wi-Fi Protected Setup to add a new wireless network profile.
Wireless Status Wi-Fi Profile
Contains information about the wireless network. Contains up to 3 wireless profiles for the phone. Includes a wireless profile for the Cisco Unified Communications Server by default.
Bluetooth
For more information on configuring Bluetooth, see Configuring Bluetooth (Cisco SPA525G/525G2 only), page 77.
Parameter
Bluetooth Device Bluetooth Status (Cisco SPA525G2 only)
Description
Click On to enable Bluetooth.
Shows the name and status of any connected Bluetooth devices.
319

B
Description
Shows the method of Bluetooth connection chosen:
Parameter
Bluetooth Mode (Cisco SPA525G2 only)
PhoneYour Cisco SPA525G2 will pair with a Bluetooth headset only. Choose this option if you will not use the Cisco SPA525G2 with a Bluetooth-enabled mobile phone. HandsfreeYour Cisco SPA525G2 will operate as a handsfree device with a Bluetooth-enabled mobile phone. BothYour Cisco SPA525G2 will use a Bluetooth headset, or operate with your Bluetooth-enabled mobile phone (see Pairing Your Cisco SPA525G2 with a Bluetooth-Enabled Mobile Phone, page 68). Note that your Cisco SPA525G2 will connect to only one device at a time (either the Bluetooth headset or the Bluetooth-enabled mobile phone.)
Bluetooth Profiles (Cisco SPA525G2 only)
This table shows the MAC (hardware) address, device name, and other information about each Bluetooth device that is associated with the Cisco SPA525G/525G2. If multiple Bluetooth devices are in range of the Cisco SPA525G/G2, the order of devices in the order list is used, and the device with a higher priority is activated first. Click the arrow keys to move devices up and down in priority. You can choose yes or no to determine if the phone connects automatically if it detects the Bluetooth device. You can also remove devices in this window.
Bluetooth Device List (Cisco SPA525G2 only)
Press Scan for Bluetooth Devices to locate Bluetooth devices in the area. Found devices are shown with the type of device, MAC address, and device name.
Personal Address Book

Address book for the phone. For more information, see the Cisco Small Business IP
Phone SPA525G/525G2 User Guide.
320

Call History
Displays the call history for the phone. To change the information displayed, select the type of call history from the drop-down list: All Calls Received Calls Placed Calls Missed Calls
Speed Dials
See Speed Dial, page 312.
Firmware Upgrade
Used to upgrade the firmware for the Cisco SPA525G/525G2. See Upgrading Firmware, page 18.
321
C
Where to Go From Here
Cisco provides a wide range of resources to help you and your customer obtain the full benefits of the SPA IP phone. Support Cisco Small Business Support Community Cisco Small Business Support and Resources Phone Support Contacts www.cisco.com/go/smallbizsupport www.cisco.com/go/smallbizhelp www.cisco.com/en/US/support/ tsd_cisco_small_business _support_center_contacts.html www.cisco.com/cisco/web/download/ index.html Select a link to download firmware for Cisco Small Business Products. No login is required.
Cisco Small Business Firmware Downloads
Product Documentation
Cisco SPA300 Series IP Phones Cisco SPA500 Series IP Phones
www.cisco.com/go/300phones www.cisco.com/go/spa500phones www.cisco.com/en/US/partner/products/ps10033/ tsd_products_support_series_home.html www.cisco.com/en/US/products/ps10042/ tsd_products_support_series_home.html www.cisco.com/en/US/products/ps10030/ tsd_products_support_series_home.html www.cisco.com/en/US/products/ps7293/ tsd_products_support_series_home.html
Cisco WIP310 IP Phone Accessories Cisco SPA 9000 Voice System Cisco Unified Communications 500 Series for Small Business
322
Where to Go From Here
C
Cisco Small Business Cisco Partner Central for Small Business (Partner Login Required) Cisco Small Business Home www.cisco.com/web/partners/sell/smb
www.cisco.com/smb
NOTE For older Cisco IP phone models, such as the Cisco SPA9XX, see the
Cisco SPA9XX Phone Administration Guide on cisco.com. This guide covers

only the Cisco SPA300 Series IP phones, Cisco SPA500 Series IP phones, and the Cisco WIP310.
323

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ADMINISTRATION GUIDE

Document revised December 2010

2010 Cisco Systems, Inc. All rights reserved.

Chapter 1: Getting Started

Prerequisites Upgrading Firmware

Using the Web-Based Configuration Utility

Chapter 2: Configuring Lines and Extensions

Chapter 3: Customizing Cisco SPA and Wireless IP Phones

Configuring Linksys Key System Parameters Enabling Call Features

Enabling Paging (Intercom)

Enabling Secure Call Enabling Service Announcements

Customizing Phone Softkeys

Configuring the Message Waiting Indicator Configuring Ring Tones

Chapter 4: Configuring SIP, SPCP, and NAT

RFC3311 Support Support for SIP NOTIFY XML-Service

122 123 123

124 124 124 125 126

Chapter 5: Configuring Security, Quality, and Network Features

Ensuring Voice Quality

Configuring Voice Codecs Configuring Domain and Internet Settings

Setting Optional Network Parameters Configuring VLAN Settings for IP Phones

Configuring the VLAN Settings

Configuring SSL VPN on the Cisco SPA525G/525G2

Chapter 6: Provisioning Basics

IP Phone Configuration Profiles

Sample Configuration File Upgrading, Resyncing, and Rebooting Phones

169 169 170 171

Chapter 7: Configuring Regional Parameters and Supplementary Services

Call Progress Tones

179 180 181

Localizing Your IP Phone

Selecting a Display Language Creating a Dictionary Server Script

Chapter 8: Configuring Dial Plans

Interdigit Long Timer (Incomplete Entry Timer)

Interdigit Short Timer (Complete Entry Timer)

Editing Dial Plans on the IP Phone Resetting the Control Timers

Chapter 9: Configuring the Cisco SPA500S Attendant Console

Configuring BroadSoft Busy Lamp Field Auto-Configuration Attendant Console Parameters

Monitoring the Cisco SPA500S

Chapter A: Creating an LED Script

Appendix B: Cisco SPA IP Phone Field Reference

235 236 236 237 239 239 239

Provisioning Tab Regional Tab

283 284 285 286 286 288 291 291 292

Attendant Console Tab (Cisco SPA500 Series)

Attendant Console Status Cisco SPA525G/525G2-Specific Tabs

Appendix C: Where to Go From Here

Overview of the Phones

4 fixed (forward, cancel, conference, and transfer) 4 dynamic

SPA502G SPA504G SPA508G SPA509G SPA525G/ SPA525G2

128 X 64 monochrome LCD with backlight

Four-way navigation key

Four-way navigation key with centerselect button

Cisco SPA500S Attendant Console

Wireless Router Analog Phone Lines Integrated Access Device

Analog Phones or Fax Machines

Figure 1 Network Configuration Example

Cisco SPA 9000 Voice System

Voice System Administration Guide.

Cisco Unified Communications 500 Series for Small Business