ADMINISTRATION GUIDE Cisco SPA232D Mobility Enhanced Phone Adapter
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Contents Chapter 1: Getting Started Feature Overview 7 7 Understanding Voice Service Operations 8 ATA Voice Features 8 Before You Begin 14 Product Features 14 Connecting the Equipment 17 Configuration and Management of the ATA 18 Registering a Cisco SPA302D Handset 19 Additional Information 20 Using the IVR for Administration 20 Mounting the ATA 24 Elements of the User Interface 26 Chapter 2: Quick Setup for Voice over IP Service 28 Chapter 3: Configuring the Network 31 Basic Se
Contents Manually Adding Port Forwarding 48 DMZ 50 Chapter 4: Configuring the Voice Settings 51 Information 52 System 62 SIP 65 Provisioning 75 Regional 81 Line 1 Settings (PHONE Port) 99 PSTN (LINE Port) 118 User 1 139 PSTN User 144 DECT Line 1 - DECT Line 10 146 DECT User 166 Chapter 5: Administration Settings Management 168 168 Web Access Management 169 TR-069 171 SNMP 173 User List (Password Management) 175 Bonjour 176 Reset Button 176 Logging 176 Log Module
Contents Factory Defaults 183 Firmware Upgrade 183 Configuration Management 184 Backup Configuration 184 Restore Configuration 184 Reboot 185 Chapter 6: Viewing the Status and Statistics 186 System Information 186 Interface Information 187 Internet Status 188 Port Statistics 189 DHCP Server Information 190 Appendix A: Frequently Asked Questions 192 Appendix B: Using the IVR for Administration 195 Appendix C: Advanced Options for Voice Services 200 Optimizing Fax Completion Rat
Contents VoIP to PSTN Call With and Without Authentication 210 Call Forwarding to PSTN Gateway 212 Configuring Dial Plans 213 Digit Sequences 214 Acceptance and Transmission of the Dialed Digits 218 Dial Plan Timer (Off-Hook Timer) 219 Interdigit Long Timer (Incomplete Entry Timer) 220 Interdigit Short Timer (Complete Entry Timer) 221 Resetting the Control Timers 221 Appendix D: Where to Go From Here Cisco SPA232D Administration Guide 223 6
1 Getting Started Thank you for choosing the Cisco SPA232D Mobility Enhanced Analog Telephone Adapter (ATA). This chapter provides more information about the features of the product and provides instructions about connecting the equipment and getting started in the web-based configuration utility.
1 Getting Started Feature Overview Understanding Voice Service Operations The ATA allows calls to be made by using SIP-based Voice-over-IP (VoIP) services and traditional telephone Public Switched Telephone Network (PSTN) services. Calls can be placed and received by using an analog phone or fax machine and Cisco SPA302D handsets.
1 Getting Started Feature Overview Supported Codecs The ATA supports the codecs listed below. You can use the default settings or configure the codec settings in the Audio Configuration section of these pages: Line 1 Settings (PHONE Port), PSTN (LINE Port), and DECT Line 1 - DECT Line 10. Codec Description G.711 (A-law and mu-law) Very low complexity codecs that support uncompressed 64 kbps digitized voice transmissions at one through ten 5 ms voice frames per packet.
1 Getting Started Feature Overview SIP Proxy Redundancy In typical commercial IP Telephony deployments, all calls are established through a SIP proxy server. A typical SIP proxy server can handle thousands of subscribers. It is important that a backup server be available so that an active server can be temporarily switched out for maintenance. The ATA supports the use of backup SIP proxy servers (through DNS SRV) so that service disruption is minimized.
1 Getting Started Feature Overview - FAX pass-through mode is triggered by the detection of a CED/CNG tone or an NSE event. - Echo canceller is automatically disabled for Modem passthrough mode. - Echo canceller is disabled for FAX pass-through if the parameter FAX Disable ECAN (Line 1 or 2 tab) is set to “yes” for that line (in that case FAX pass-through is the same as Modem pass-through) - Call waiting and silence suppression are automatically disabled for both FAX and Modem pass-through.
1 Getting Started Feature Overview information. You can configure this setting in the RTP Parameters section of the SIP page. • Call Progress Tones The ATA has configurable call progress tones. Call progress tones are generated locally on the ATA so that an end user is advised of status (such as ringback) Parameters for each type of tone (for instance a dial tone played back to an end user) may include frequency and amplitude of each component, and cadence information.
1 Getting Started Feature Overview string, and a short timeout, signaling that more digits are expected. For more information, see Configuring Dial Plans, page 213. • Polarity Control The ATA allows the polarity to be set when a call is connected and when a call is disconnected. This feature is required to support some pay phone system and answering machines. You can configure these settings in the FXS Port Polarity Configuration section of the Line 1 Settings (PHONE Port) page.
1 Getting Started Before You Begin Before You Begin Before you begin the installation, make sure that you have the following equipment and services: • An active Internet account and Voice over IP account • Ethernet cable to connect to your broadband network device • Phone to connect to your ATA • Phone cable to connect your phone • Optional: Uninterruptible Power Supply (UPS) to provide backup power • Optional: Cisco SPA302D Mobility Enhanced Cordless Handsets Product Features Top Panel Featur
1 Getting Started Product Features Feature SYSTEM Description Steady green—The system is ready. Slow flashing green—Acquiring an IP address, if applicable. (DHCP is used by default.) Fast flashing green—Upgrading the firmware. Off—There is no power or the system cannot boot up. Back Panel Feature Description RESET Performs two functions: Restart the ATA: Press quickly (less than a second) with a paperclip or similar object. Restore the factory default settings: Press and hold for 5 to s6 seconds.
1 Getting Started Product Features Default Settings Parameter Administrator Username Administrator Password User Username User Password Internet Connection Type LAN IP Address (Also the address for the webbased configuration utility.) DHCP Range (DHCP server enabled by default.) Netmask PIN for handset registration, IP settings, and SIP settings Cisco SPA232D Administration Guide Default Value admin admin cisco cisco Automatic Configuration - DHCP 192.168.15.1 192.168.15.100-149 255.255.255.
1 Getting Started Connecting the Equipment Connecting the Equipment NOTE For wall-mounting instructions, see Additional Information, page 20. STEP 1 Connect the provided Ethernet cable to the INTERNET (Blue) port. Connect the other end of the cable directly to your broadband network device. STEP 2 Connect the provided phone cable to the PHONE 1 (Gray) port. Connect the other end of the cable to your analog phone or fax machine.
Getting Started Configuration and Management of the ATA 1 Configuration and Management of the ATA You can use the web-based configuration utility to set up your ATA. You also can use the built-in Interactive Voice Response (IVR) system. (See Using the IVR for Administration, page 20.) STEP 1 Connect the provided Ethernet network cable to the ETHERNET (Yellow) port of the ATA. Connect the other end of the cable to the Ethernet port of your PC. STEP 2 Power on your computer.
Getting Started Registering a Cisco SPA302D Handset 1 Note: The Cisco SPA232D assigns DECT Line1 as the default line for outgoing calls from Cisco SPA302D handsets. If needed, you can configure additional VoIP accounts as separate “DECT Lines.” To do so, choose the Voice menu, and then use the DECT Line 1~10 links in the navigation tree. Use the check boxes on the Quick Setup page to associate the DECT Line(s) to each handset. STEP 7 Click Submit to save your settings.
1 Getting Started Additional Information Additional Information Using the IVR for Administration An IVR system is available to help you to configure and manage your ATA. You can use the telephone keypad to select options and to make your entries. To access the IVR menu: STEP 1 Connect an analog phone to the PHONE port of the ATA. STEP 2 Press the star key four times: **** STEP 3 After the greeting plays, press the keys on the phone keypad to select your options.
1 Getting Started Additional Information • To enter the decimal points in an IP address, press the * (star) key. For example, to enter the IP address 191.168.1.105, perform the following tasks: –Press these keys: 191*168*1*105. –Press the # (pound) key to indicate that you have finished entering the IP address. –Press 1 to save the IP address or press the * (star) key to cancel your entry and return to the main menu.
1 Getting Started Additional Information IVR Action Set Network Mask Menu Option 121 Choices and Instructions To enter the value, press numbers on the telephone key pad. Press the * (star) key to enter a decimal point. Note: This option is available only after you choose Static IP as the Internet Connection Type, through option 101. Check Gateway IP Address Set Gateway IP Address 130 131 To enter the value, press numbers on the telephone key pad. Press the * (star) key to enter a decimal point.
1 Getting Started Additional Information IVR Action Menu Option SPA122 only: Check LAN 210 IP address (Ethernet port) Announce Line 1 SIP 1910 Transport Set Line 1 SIP Transport 1911 Check Line 2 SIP Transport Set Line 2 SIP Transport 1920 Exit IVR Allow or prevent WAN access to the administration web server 3948 7932 1921 The system will allow WAN access only if the default admin username and password have been changed in the Configuration Utility.
1 Getting Started Additional Information Mounting the ATA You can place the ATA on a desktop or mount it on a wall. ! CAUTION To prevent the ATA from overheating, do not operate it in an area that exceeds an ambient temperature of 104°F (40°C). Desktop Placement Place the ATA on a flat surface near an electrical outlet. WARNING Do not place anything on top of the ATA; excessive weight could damage it. Wall Mounting The ATA has two wall-mount slots on the bottom panel.
1 Getting Started Additional Information To mount the unit to the wall: STEP 1 Determine where you want to mount the unit. Verify that the surface is smooth, flat, dry, and sturdy. STEP 2 Drill two pilot holes into the surface 58 mm apart (about 2.28 in.). Make sure that the holes are at the same height above the floor so that the unit is level and secure in either of its two safety-certified orientations. STEP 3 Insert a screw into each hole, leaving a gap of 5 mm (0.1968 in.
1 Getting Started Elements of the User Interface Elements of the User Interface Before you use your ATA, become familiar with the following features of the user interface. Screen Elements 1 3 2 Component Description 1. Menu Bar (top) Provides access to the modules of the configuration utility. Click a menu to view the options in the navigation tree. 2. Navigation Tree (left panel) Provides access to the configuration pages within the selected module.
1 Getting Started Elements of the User Interface Configuration Utility Icons Many configuration pages provide the following icons for common tasks. Icon Description Edit Icon The Edit icon lets you edit an existing item from a list. After making your changes, click the Submit button to save your changes. Add Item Icon The Add Item icon lets you add an item to a list. After you have created a new item, click the Submit button to save the new item.
2 Quick Setup for Voice over IP Service The Quick Setup page is displayed automatically when you first log on ATA. You can use this page to quickly configure connectivity to your provider’s Voice over IP network for your analog phone and Cisco SPA302D handsets. NOTE Connecting to your service provider’s network requires Internet connectivity.
Quick Setup for Voice over IP Service 2 • Password: Enter the password that is required to log in to your Internet account. • Dial Plan in (Line section only): Keep the default settings (recommended) or edit the dial plan to suit your site. For more information, see Configuring Dial Plans, page 213. STEP 2 DECT Handset Outgoing Line Selection: For each DECT Handset, check the boxes to choose the DECT Lines for outgoing calls. Uncheck the boxes for the lines that you do not want to use.
Quick Setup for Voice over IP Service 2 b. Use an external phone to place an inbound call to the telephone number that was assigned by your ITSP. Verify that the phone rings and you have two-way audio on the call.
3 Configuring the Network This chapter describes how to configure the network settings for your ATA. It includes the following sections: • Basic Setup • Advanced Settings • Application Basic Setup Use the Network Setup > Basic Setup pages to configure your Internet connection, local network settings, and your time settings.
3 Configuring the Network Basic Setup You can configure the ATA to operate in one of the following modes: • NAT: Network Address Translation (NAT) is a function that allows multiple devices on a private network to share a public, routable IP address to establish connections over the Internet. To enable Voice over IP service to co-exist with NAT, some form of NAT traversal is required, either on the ATA or another network device. Use this option if your ATA connects to one network on the WAN port (10.0.0.
3 Configuring the Network Basic Setup Internet Settings Use the Network Setup > Basic Setup > Internet Settings page to set up your Internet connection. To open this page: Click Network in the menu bar, and then click Basic Setup > Internet Settings in the navigation tree. Enter the settings as described in the table. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.
3 Configuring the Network Basic Setup Field Static IP Settings Description • Internet IP Address and Subnet Mask: Enter the IP address and subnet mask that was assigned to your account by your service provider. This address is seen by external users on the Internet. • Default Gateway: Enter the Gateway IP Address that was provided by your ISP. If needed, you can adjust the MTU and Optional Settings, as described below.
3 Configuring the Network Basic Setup Field Description MTU The Maximum Transmission Unit (MTU) setting specifies the largest protocol data unit (in bytes) permitted for network transmission. Generally, a larger MTU means greater efficiency. However, a larger packet may cause delays for other traffic and is more likely to become corrupted. In most cases, you should keep the default setting, Auto, to allow the ATA to choose the appropriate MTU.
3 Configuring the Network Basic Setup Feature Description Secondary DNS Enter the IP address of the secondary Domain Name Service (DNS) server to use for domain name resolution. Keep the default entry, 0.0.0.0, to use the secondary DNS server that is specified for the WAN connection. Network Settings for the LAN and DHCP Server Use the Network Setup > Basic Setup > Network Settings page to set the IP address and subnet mask for your local network.
3 Configuring the Network Basic Setup Field Description IP Reservation: Click the Show DHCP Reservation button to view and manage the DHCP client list. Click the Hide DHCP Reservation button to hide the list. When the list is displayed, you can perform the following tasks: • To reserve a static IP address for a current DHCP client: Check the box for the client in the Select Clients from DHCP Tables list. Click Add Clients. The selected clients are added to the Clients Already Reserved list.
3 Configuring the Network Basic Setup Field Description Client Lease Time Enter the number of minutes that a dynamically assigned IP address can be in use, or “leased.” After this time elapses, a client device has to request a DHCP lease renewal. Use 0 to represent 1 day, 9999 never expire. Default setting: 0 Option 66 Provides provisioning server address information to hosts that request this option.
3 Configuring the Network Basic Setup Field Description Option 159 Provides a configuration URL to clients that request this option. An option 159 URL defines the protocol and path information by using an IP address for clients that cannot use DNS. For example: https://10.1.1.1:888/configs/bootstrap.cfg Default setting: blank Option 160 Provides a configuration URL to clients that request this option.
3 Configuring the Network Basic Setup User Manual If you prefer to set the system manually rather than automatically obtaining the settings from an NTP server, click User Manual and then enter the date and time. Field Description Date Enter the date in the following order: four-digit year, month, day. Time Enter the time in the following order: hour (from 1 to 24), minutes, and seconds. Time Zone To use a time server to establish the time settings, select Time Zone.
3 Configuring the Network Advanced Settings Field Description Auto Recovery After Reboot Choose this option to allow the ATA to automatically reconnect to the time server after a system reboot. Default setting: Disabled Advanced Settings Use the Network Setup > Advanced Settings pages to configure features including port flow control, MAC address cloning, VPN passthrough, and VLAN.
3 Configuring the Network Advanced Settings Field Description Flow Control Flow control is a mechanism that temporarily stops the transmission of data on a port. For example, a situation might arise where a sending station (computer) is transmitting data faster than some other part of the network (including the receiving station) can accept. The overwhelmed network element will halt the transmission of the sender for a specified period of time.
3 Configuring the Network Advanced Settings Field Description MAC Clone Click Enabled to enable MAC address cloning, or click Disabled to disable this feature. Default setting: Disabled. MAC Address Enter the MAC address that you want to assign to your ATA. If your computer’s MAC address is the address that you previously registered for your ISP account, click Clone Your PC’s MAC. Your computer’s MAC address appears in the MAC Address field.
3 Configuring the Network Advanced Settings Field Description L2TP Passthrough Layer 2 Tunneling Protocol is the method used to enable Point-to-Point sessions via the Internet on the Layer 2 level. Click Enabled to enable this feature, or click Disabled to disable it. Default setting: Enabled VLAN Use the Network Setup > Advanced Settings > VLAN page to assign a VLAN ID to your network. For example, your call control system may require a particular voice VLAN ID.
3 Configuring the Network Advanced Settings CDP & LLDP Device discovery protocols enable directly connected devices to discover information about each other. You may wish to enable these protocols to allow your network management system to learn about your ATA and endpoints. Use the Network Setup > Advanced Settings > CDP & LLDP page to specify the settings for Cisco Discovery Protocol (CDP) and the Link Layer Discovery Protocol (LLDP).
3 Configuring the Network Application Application Use the Network Setup > Application pages to support voice service and any servers that you host for public access. • Quality of Service (QoS) • Port Forwarding • DMZ Quality of Service (QoS) Use the Network Setup > Application > QoS page to set the upstream bandwidth to suit your broadband service. This feature is enabled by default and helps to ensure that voice is prioritized during periods of heavy network traffic.
3 Configuring the Network Application Port Forwarding Use the Network Setup > Application > Port Forwarding page if you need to explicitly allow access to specific ports from external devices. To open this page: Click Network Setup in the menu bar, and then click Application > Port Forwarding in the navigation tree. List of Port Forwarding To add a port forwarding rule, click Add Entry. To edit a port forwarding rule, select it in the list and then click the pencil icon.
3 Configuring the Network Application Manually Adding Port Forwarding Use this page to enter the port forwarding settings for an application. To open this page: On the Network Setup > Application > Port Forwarding page, click the Add Entry button or the pencil icon. Enter the settings as described below. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.
3 Configuring the Network Application Field Description External Port, Internal Port For Single Port Forwarding, specify the ports to use. For simplicity, the internal and external port numbers will often be the same. However, different external port numbers could be used to differentiate traffic of the same application type intended for different internal servers, or to promote privacy through the use of nonstandard ports.
3 Configuring the Network Application DMZ Use the Network Setup > Application > DMZ page if you need to allow a local device to be exposed to the Internet for a special-purpose service. The specified network device must have its DHCP client function disabled and must have a reserved IP address (also known as a static IP address) to ensure that it is reachable at the specified IP address. To reserve an IP address for a device, see Network Settings for the LAN and DHCP Server, page 36.
4 Configuring the Voice Settings This chapter describes how to configure the voice settings and voice services for the ATA. It includes the following sections: • Information • System • SIP • Provisioning • Regional • Line 1 Settings (PHONE Port) • PSTN (LINE Port) • User 1 • PSTN User • DECT Line 1 - DECT Line 10 • DECT User NOTE For additional information, see Appendix C, “Advanced Options for Voice Services.
4 Configuring the Voice Settings Information Information Use the Voice > Information page to view information about the ATA voice application. To open this page: Click Voice on the menu bar, and then click Information in the navigation tree. Enter the settings as described below. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings. Product Information Field Description Product Name Model number/name.
4 Configuring the Voice Settings Information Field Description Elapsed Time Total time elapsed since the last reboot of the system; for example, 25 days and 18:12:36. RTP Packets Sent Total number of RTP packets sent (including redundant packets) RTP Bytes Sent Total number of RTP bytes sent. RTP Packets Recv Total number of RTP packets received (including redundant packets) RTP Bytes Recv Total number of RTP bytes received.
4 Configuring the Voice Settings Information Field Description Call Back Active Indicates whether or not a call back request is in progress. Options are either yes or no. Last Called Number The phone number that was most recently called through this port. Last Caller Number The originating phone number of the call that was most recently received through this port.
4 Configuring the Voice Settings Information Field Description Call 1 and 2 Type The direction of the call.
4 Configuring the Voice Settings Information Field Description Call 1 and 2 Decode Latency The number of milliseconds for decoder latency. Call 1 and 2 Jitter The number of milliseconds for receiver jitter Call 1 and 2 Round Trip Delay The number of milliseconds for delay. Call 1 and 2 Packets Lost The number of packets lost. Call 1 and 2 Packet Error The number of invalid packets received.
4 Configuring the Voice Settings Information Field Description Last Called VoIP Number The VoIP phone number that was most recently called through this port. Last Called PSTN Number The PSTN phone number that was most recently called through the LINE port Last VoIP Caller The originating phone number of the VoIP call that was most recently received through the LINE port. Last PSTN Caller The originating phone number of the PSTN call that was most recently received through the LINE port.
4 Configuring the Voice Settings Information Field Description Call Type The type of call: VoIP State • PSTN Gateway Call = VoIP-To-PSTN Call • VoIP Gateway Call = PSTN-To-VoIP Call • PSTN To Line 1 = PSTN call ring through and answered by Line 1 • Line 1 Forward to PSTN Gateway = VoIP calls Line 1 then forwarded to PSTN GW • Line 1 Forward to PSTN Number =VoIP calls Line 1 then forwarded to PSTN number • Line 1 To PSTN Gateway • Line 1 Fallback To PSTN Gateway May take one of the follo
4 Configuring the Voice Settings Information Field Description PSTN Peer Name The name of the party at the PSTN call leg. VoIP Peer Number The phone number of the party at the VoIP call leg. PSTN Peer Number The phone number of the party at the PSTN call leg. VoIP Call Encoder The audio encoder being used for the VoIP call leg. VoIP Call Decoder The audio decoder being used for the VoIP call leg. VoIP Call FAX The status of the fax passthrough mode for VoIP calls.
4 Configuring the Voice Settings Information DECT 1 ~ DECT 10 Status Field Description Registration State Indicates whether or not the line has registered with the SIP proxy: Registered, Not Registered, or Failed. Last Registration At The last date and time when the line was registered. Next Registration In The number of seconds before the next registration renewal. Message Waiting Indicates whether or not there are new messages: yes or no.
4 Configuring the Voice Settings Information Field Description Call 1 and 2 FAX The status of the fax passthrough mode.
4 Configuring the Voice Settings System Field Description Call 1 and 2 Decode Latency The number of milliseconds for decoder latency. Call 1 and 2 Jitter The number of milliseconds for receiver jitter Call 1 and 2 Round Trip Delay The number of milliseconds for delay. Call 1 and 2 Packets Lost The number of packets lost. Call 1 and 2 Packet Error The number of invalid packets received. Call 1 and 2 Mapped RTP Port The port mapped for Real Time Protocol traffic for Call 1/2.
4 Configuring the Voice Settings System • You can deploy a syslog server to receive syslog messages from the ATA, which acts as a syslog client. The syslog client device uses the syslog protocol to send messages, based on its configuration, to a syslog server. The syslog messages can be accessed by reviewing the "syslog.514.log" file which resides in the same directory as the slogsrv.exe syslog server application.
4 Configuring the Voice Settings System Field Description 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. Default setting: blank Debug Level Determines the level of debug information that will be generated. Select 0, 1, 2, 3 or 3+Router from the dropdown list. The higher the debug level, the more debug information will be generated.
4 Configuring the Voice Settings SIP SIP Use the Voice > SIP page to configure SIP parameters and values. To open this page: Click Voice on the menu bar, and then click SIP in the navigation tree. Enter the settings as described below. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings. NOTE For a deeper understanding of these fields, refer to Request for Comments (RFC) 3261.
4 Configuring the Voice Settings SIP Field Description SIP Accept Language Accept-Language header used. There is no default (this indicates that the ATA does not include this header) If empty, the header is not included. Default setting: blank DTMF Relay MIME Type The MIME Type used in a SIP INFO message to signal a DTMF event. Default setting: application/dtmf-relay. Hook Flash MIME Type The MIME Type used in a SIP INFO message to signal a hook flash event.
4 Configuring the Voice Settings SIP Field Description RFC 2543 Call Hold Configures the type of call hold: a:sendonly or 0.0.0.0. Do not use the 0.0.0.0 syntax in a HOLD SDP; use the a:sendonly syntax. Default setting: no Mark all AVT Packets Select yes if you want all AVT tone packets (encoded for redundancy) to have the marker bit set for each DTMF event. Select no to have the marker bit set only for the first packet.
4 Configuring the Voice Settings SIP Field Description SIP Timer F Non-INVITE time-out value, which can range from 0 to 64 seconds. Default setting: 32 SIP Timer H H INVITE final response, time-out value, which can range from 0 to 64 seconds. Default setting: 32 SIP Timer D ACK hang-around time, which can range from 0 to 64 seconds. Default setting: 32 SIP Timer J Non-INVITE response hang-around time, which can range from 0 to 64 seconds.
4 Configuring the Voice Settings SIP Field Description Reg Retry Long Intvl When registration fails with a SIP response code that does not match Retry Reg RSC, the ATA waits for the specified length of time before retrying. If this interval is 0, the ATA stops trying. This value should be much larger than the Reg Retry Intvl value, which should not be 0. Default setting: 1200 Reg Retry Random Delay Random delay range (in seconds) to add to Register Retry Intvl when retrying REGISTER after a failure.
4 Configuring the Voice Settings SIP Field Description SIT3 RSC SIP response status code to INVITE on which to play the SIT3 Tone. Default setting: blank SIT4 RSC SIP response status code to INVITE on which to play the SIT4 Tone. Default setting: blank Try Backup RSC SIP response code that retries a backup server for the current request. Default setting: blank Retry Reg RSC Interval to wait before the ATA retries registration after failing during the last registration.
4 Configuring the Voice Settings SIP Field Description RTCP Tx Interval Interval for sending out RTCP sender reports on an active connection. It can range from 0 to 255 seconds. During an active connection, the ATA can be programmed to send out compound RTCP packet on the connection. Each compound RTP packet except the last one contains a SR (Sender Report) and a SDES (Source Description) The last RTCP packet contains an additional BYE packet.
4 Configuring the Voice Settings SIP SDP Payload Types Field Description NSE Dynamic Payload NSE dynamic payload type. The valid range is 96-127. Default setting: 100 AVT Dynamic Payload AVT dynamic payload type. The valid range is 96-127. Default setting: 101 INFOREQ Dynamic Payload INFOREQ dynamic payload type. Default setting: blank G726r32 Dynamic Payload G726r32 dynamic payload type. Default setting: 2 G729b Dynamic Payload G.729b dynamic payload type. The valid range is 96-127.
4 Configuring the Voice Settings SIP Field Description G729b Codec Name G.729b codec name used in SDP. Default setting: G729ab EncapRTP Codec Name EncapRTP codec name used in SDP. Default setting: encaprtp NAT Support Parameters Field Description Handle VIA received If you select yes, the ATA processes the received parameter in the VIA header (this value is inserted by the server in a response to any one of its requests) If you select no, the parameter is ignored.
4 Configuring the Voice Settings SIP Field Description STUN Enable Enables the use of STUN to discover NAT mapping. Select yes or no from the drop-down menu. Default setting: no STUN Test Enable If the STUN Enable feature is enabled and a valid STUN server is available, the ATA can perform a NAT-type discovery operation when it powers on. It contacts the configured STUN server, and the result of the discovery is reported in a Warning header in all subsequent REGISTER requests.
4 Configuring the Voice Settings Provisioning Linksys Key System Parameters Field Description Linksys Key System To enable operation with the Cisco SPA9000, choose yes. Otherwise, choose no. Default setting: no Multicast Address The multicast address for devices in the Cisco SPA9000 voice network. Default setting: 224.168.168.168:6061 Key System Auto Discovery To enable auto-discovery of the Cisco SPA9000 voice system, choose yes. Otherwise, choose no.
4 Configuring the Voice Settings Provisioning Field Description Resync On Reset Triggers a resync after every reboot except for reboots caused by parameter updates and firmware upgrades. Default setting: yes Resync Random Delay The maximum value for a random time interval that the ATA waits before making its initial contact with the provisioning server. This delay is effective only on the initial configuration attempt following power-on or reset.
4 Configuring the Voice Settings Provisioning Field Description Resync Error Retry Delay Resync retry interval (in seconds) applied in case of resync failure. The ATA has an error retry timer that activates if the previous attempt to sync with the provisioning server fails. The ATA waits to contact the server again until the timer counts down to zero. This parameter is the value that is initially loaded into the error retry timer.
4 Configuring the Voice Settings Provisioning Field Description Resync Fails On FNF Determines whether a file-not-found response from the provisioning server constitutes a successful or a failed resync. A failed resync activates the error resync timer. Default setting: yes Profile Rule This parameter is a profile script that evaluates to the provisioning resync command. The command is a TCP/IP operation and an associated URL. The TCP/IP operation can be TFTP, HTTP, or HTTPS.
4 Configuring the Voice Settings Provisioning Field Description Report Rule The target URL to which configuration reports are sent. This parameter has the same syntax as the Profile_Rule parameter, and resolves to a TCP/IP command with an associated URL. A configuration report is generated in response to an authenticated SIP NOTIFY message, with Event: report. The report is an XML file containing the name and value of all the device parameters. This parameter may optionally contain an encryption key.
4 Configuring the Voice Settings Provisioning Field Description Log Upgrade Request Msg Syslog message issued at the start of a firmware upgrade attempt. Default setting: $PN $MAC -- Requesting upgrade $SCHEME://$SERVIP:$PORT$PATH Log Upgrade Success Msg Syslog message issued after a firmware upgrade attempt completes successfully.
Configuring the Voice Settings Regional 4 Regional Use the Voice > Regional page to localize your system with the appropriate regional settings. To open this page: Click Voice on the menu bar, and then click Region in the navigation tree. Enter the settings as described below. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.
Configuring the Voice Settings Regional 4 Segment 3: On=0.2s, Off=0.2s Segment 4: On=1.0s, Off=4.0s Total Ring Length = 60s FreqScript A mini-script of up to 127 characters that specifics the frequency and level parameters of a tone.
4 Configuring the Voice Settings Regional Number of Frequencies = 2 Frequency 1 = 350 Hz at –19 dBm Frequency 2 = 440 Hz at –19 dBm Number of Cadence Sections = 2 Cadence Section 1: Section Length = 2s Number of Segments = 1 Segment 1: On=0.1s, Off=0.1s with Frequencies 1 and 2 Cadence Section 2: Section Length = 10s Number of Segments = 1 Segment 1: On=forever, with Frequencies 1 and 2 Total Tone Length = 12s Enter the settings as described below.
4 Configuring the Voice Settings Regional Field Description Off Hook Warning Tone Played when the caller has not properly placed the handset on the cradle. Off Hook Warning Tone is played when the Reorder Tone times out. Default setting: 480@-3,620@3;10(.125/.125/1+2) Ring Back Tone Played during an outbound call when the far end is ringing.
4 Configuring the Voice Settings Regional Field Description MWI Dial Tone Played instead of the Dial Tone when there are unheard messages in the caller’s mailbox. Default setting: 350@-5,440@-5;2(.1/.1/1+2);10(*/0/1+2) Cfwd Dial Tone Played when all calls are forwarded. Default setting: 350@-5,440@-5;2(.2/.2/1+2);10(*/0/1+2) Holding Tone Informs the local caller that the far end has placed the call on hold. Default setting: 600@-5;*(.1/.1/1,.1/.1/1,.1/9.
4 Configuring the Voice Settings Regional Field Description Ring4 Cadence Cadence script for distinctive ring 4. Default setting: 60(.3/.2,1/.2,.3/4) Ring5 Cadence Cadence script for distinctive ring 5. Default setting: 1(.5/.5) Ring6 Cadence Cadence script for distinctive ring 6. Default setting: 60(.2/.4,.2/.4,.2/4) Ring7 Cadence Cadence script for distinctive ring 7. Default setting: 60(.4/.2,.4/.2,.4/4) Ring8 Cadence Cadence script for distinctive ring 8. Default setting: 60(0.25/9.
4 Configuring the Voice Settings Regional Distinctive Ring/CWT Pattern Names Field Description Ring1 Name Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 1 for the inbound call. Default setting: Bellcore-r1 Ring2 Name Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 2 for the inbound call. Default setting: Bellcore-r2 Ring3 Name Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 3 for the inbound call.
4 Configuring the Voice Settings Regional - Ring Frequency: 25 - Ring Voltage: 80Vc Field Description Ring Waveform Waveform for the ringing signal. Choices are Sinusoid or Trapezoid. Default setting: Sinusoid Ring Frequency Frequency of the ringing signal. Valid values are 10–100 (Hz) Default setting: 20 Ring Voltage Ringing voltage. Choices are 60–90 (V) Default setting: 85 CWT Frequency Frequency script of the call waiting tone. All distinctive CWTs are based on this tone.
4 Configuring the Voice Settings Regional Field Description Reorder Delay Delay after far end hangs up before reorder tone is played. 0 = plays immediately, inf = never plays. Range: 0–255 seconds. Default setting: 5. Call Back Expires Expiration time in seconds of a call back activation. Range: 0–65535 seconds. Default setting: 1800 Call Back Retry Intvl Call back retry interval in seconds. Range: 0–255 seconds.
4 Configuring the Voice Settings Regional Field Description CPC Delay Delay in seconds after caller hangs up when the ATA starts removing the tip-and-ring voltage to the attached equipment of the called party. The range is 0–255 seconds.
4 Configuring the Voice Settings Regional Field Description Call Back Act Code Starts a callback when the last outbound call is not busy. Default setting: *66 Call Back Deact Code Cancels a callback. Default setting: *86 Call Back Busy Act Code Starts a callback when the last outbound call is busy. Default setting: *05 Cfwd All Act Code Forwards all calls to the extension specified after the activation code. Default setting: *72 Cfwd All Deact Code Cancels call forwarding of all calls.
4 Configuring the Voice Settings Regional Field Description Accept Last Act Code Accepts the last outbound call. It lets the call ring through when do not disturb or call forwarding of all calls are enabled. Default setting: *64 Accept Last Deact Code Cancels the code to accept the last outbound call. Default setting: *84 CW Act Code Enables call waiting on all calls. Default setting: *56 CW Deact Code Disables call waiting on all calls.
4 Configuring the Voice Settings Regional Field Description CID Deact Code Disables caller ID generation. Default setting: *85 CWCID Act Code Enables call waiting, caller ID generation. Default setting: *25 CWCID Deact Code Disables call waiting, caller ID generation. Default setting: *45 Dist Ring Act Code Enables the distinctive ringing feature. Default setting: *26 Dist Ring Deact Code Disables the distinctive ringing feature.
4 Configuring the Voice Settings Regional Field Description Modem Line Toggle Code Toggles the line to a modem. Modem passthrough mode can be triggered only by pre-dialing this code. Default setting: *99 FAX Line Toggle Code Toggles the line to a fax machine. Default setting: #99 Media Loopback Code Use for media loopback. Default setting: *03 Referral Services Codes These codes tell the ATA what to do when the user places the current call on hold and is listening to the second dial tone.
4 Configuring the Voice Settings Regional Field Description Feature Dial Services Codes These codes tell the ATA what to do when the user is listening to the first or second dial tone. One or more *codes can be configured into this parameter, such as *72, or *72|*74|*67|*82, etc. The maximum length is 79 characters. This parameter applies when the user has a dial tone (first or second dial tone) After receiving dial tone, a user enters the *code and the target number according to current dial plan.
4 Configuring the Voice Settings Regional Vertical Service Announcement Codes Field Description Service Annc Base Number Base number for service announcements. Default setting: blank Service Annc Extension Codes Extension codes for service announcements. Default setting: blank Outbound Call Codec Selection Codes Field Description Prefer G711u Code Dial prefix to make G.711u the preferred codec for the call. Default setting: *017110 Force G711u Code Dial prefix to make G.
4 Configuring the Voice Settings Regional Field Description Force G722 Code Dial prefix to make G.722 the only codec that can be used for the call. Default setting: *02722 Miscellaneous Field Description FXS Port Impedance Sets the electrical impedance of the PHONE port. Choices are: 600, 900, 600+2.16uF, 900+2.16uF, 270+750||150nF, 220+850||120nF, 220+820||115nF, or 200+600||100nF. Default setting: 600. NOTE For New Zealand impedance (370+620||310nF), use 270+750||150nF.
4 Configuring the Voice Settings Regional Field Description Playback ABCD To enable local playback of OOB DTMF ABCD, select yes. Otherwise, select no. Default setting: yes Caller ID Method The choices are described below. Default setting: Bellcore(N.Amer, China) • Bellcore (N.Amer,China): CID, CIDCW, and VMWI. FSK sent after first ring (same as ETSI FSK sent after first ring) (no polarity reversal or DTAS) • DTMF (Finland, Sweden): CID only.
4 Configuring the Voice Settings Line 1 Settings (PHONE Port) Field Description Caller ID FSK Standard The ATA supports bell 202 and v.23 standards for caller ID generation. Default setting: bell 202 Feature Invocation Method Select the method you want to use, Default or Sweden default. Default setting: Default Line 1 Settings (PHONE Port) Use the Voice > Line 1 page to configure the settings for calls through the PHONE port.
4 Configuring the Voice Settings Line 1 Settings (PHONE Port) Streaming Audio Server (SAS) Field Description SAS Enable To enable the use of the line as a streaming audio source, select yes. Otherwise, select no. If enabled, the line cannot be used for outgoing calls. Instead, it auto-answers incoming calls and streams audio RTP packets to the caller.
4 Configuring the Voice Settings Line 1 Settings (PHONE Port) NAT Settings Field Description NAT Mapping Enable To use externally mapped IP addresses and SIP/RTP ports in SIP messages, select yes. Otherwise, select no. Default setting: no NAT Keep Alive Enable To send the configured NAT keep alive message periodically, select yes. Otherwise, select no. Default setting: no NAT Keep Alive Msg Enter the keep alive message that should be sent periodically to maintain the current NAT mapping.
4 Configuring the Voice Settings Line 1 Settings (PHONE Port) Field Description Network Jitter Level Determines how jitter buffer size is adjusted by the ATA. Jitter buffer size is adjusted dynamically. The minimum jitter buffer size is 30 milliseconds or (10 milliseconds + current RTP frame size), whichever is larger, for all jitter level settings. However, the starting jitter buffer size value is larger for higher jitter levels.
4 Configuring the Voice Settings Line 1 Settings (PHONE Port) Field Description Auth ResyncReboot If this feature is enabled, the ATA authenticates the sender when it receives the NOTIFY resync reboot (RFC 2617) message. To use this feature, select yes. Otherwise, select no. Default setting: yes SIP Proxy-Require The SIP proxy can support a specific extension or behavior when it sees this header from the user agent.
4 Configuring the Voice Settings Line 1 Settings (PHONE Port) Field Description SIP Debug Option SIP messages are received at or sent from the proxy listen port. This feature controls which SIP messages to log. The choices are described below. Default setting: none • none—No logging. • 1-line—Logs the start-line only for all messages. • 1-line excl. OPT—Logs the start-line only for all messages except OPTIONS requests/responses. • 1-line excl.
4 Configuring the Voice Settings Line 1 Settings (PHONE Port) Field Description Referor Bye Delay The number of seconds to wait before sending a BYE to the referrer to terminate a stale call leg after a call transfer. Refer Target Bye Delay The number of seconds to wait before sending a BYE to the refer target to terminate a stale call leg after a call transfer. Referee Bye Delay The number of seconds to wait before sending a BYE to the referee to terminate a stale call leg after a call transfer.
4 Configuring the Voice Settings Line 1 Settings (PHONE Port) Call Feature Settings Field Description Blind Attn-Xfer Enable Enables the ATA to perform an attended transfer operation by ending the current call leg and performing a blind transfer of the other call leg. If this feature is disabled, the ATA performs an attended transfer operation by referring the other call leg to the current call leg while maintaining both call legs. To use this feature, select yes. Otherwise, select no.
4 Configuring the Voice Settings Line 1 Settings (PHONE Port) Field Description Emergency Number Comma separated list of emergency number patterns. If outbound call matches one of the pattern, the ATA will disable hook flash event handling. The condition is restored to normal after the call ends. Blank signifies that there is no emergency number. Maximum number length is 63 characters. Default setting: blank Mailbox ID Enter the ID number of the mailbox for this line.
4 Configuring the Voice Settings Line 1 Settings (PHONE Port) Field Description Register Expires Expires value in sec in a REGISTER request. The ATA will periodically renew registration shortly before the current registration expired. This parameter is ignored if the Register parameter is no. Range: 0 – (231 – 1) sec. Default setting: 3600 Ans Call Without Reg Allow answering inbound calls without successful (dynamic) registration by the unit.
4 Configuring the Voice Settings Line 1 Settings (PHONE Port) Field Description Mailbox Subscribe Expires Sets subscription interval for voicemail message waiting indication. When this time period expires, the ATA sends another subscribe message to the voice mail server. Default: 2147483647 Subscriber Information Field Description Display Name Display name for caller ID. Default setting: blank User ID User ID for this line. Default setting: blank Password Password for this line.
4 Configuring the Voice Settings Line 1 Settings (PHONE Port) Field Description Call Waiting Serv Enable Call Waiting Service. Default setting: yes Block CID Serv Enable Block Caller ID Service.
4 Configuring the Voice Settings Line 1 Settings (PHONE Port) Field Description Call Redial Serv Enable Call Redial Service. Call Back Serv Enable Call Back Service. Three Way Call Serv Enable Three Way Calling Service. Three Way Calling is required for Three Way Conference and Attended Transfer. Default setting: yes Three Way Conf Serv Enable Three Way Conference Service. Three Way Conference is required for Attended Transfer.
4 Configuring the Voice Settings Line 1 Settings (PHONE Port) Field Description Feature Dial Serv Enable Feature Dial Service. See the Feature Dial Services Codes parameter For more information. Default setting: yes Service Announcement Serv Enable Service Announcement Service. Default setting: no Reuse CID Number As Name Use the Caller ID number as the caller name.
4 Configuring the Voice Settings Line 1 Settings (PHONE Port) Field Description Silence Supp Enable To enable silence suppression so that silent audio frames are not transmitted, select yes. Otherwise, select no. Default setting: no G726-32 Enable To enable the use of the G.726 codec at 32 kbps, select yes. Otherwise, select no. Default setting: yes Silence Threshold Select the appropriate setting for the threshold: high, medium, or low.
4 Configuring the Voice Settings Line 1 Settings (PHONE Port) Field Description DTMF Process AVT To use the DTMF process AVT feature, select yes. Otherwise, select no. Default setting: yes FAX Process NSE To use the fax process NSE feature, select yes. Otherwise, select no. Default setting: yes DTMF Tx Method Select the method to transmit DTMF signals to the far end: InBand, AVT, INFO, or Auto. InBand sends DTMF by using the audio path. AVT sends DTMF as AVT events. INFO uses the SIP INFO method.
4 Configuring the Voice Settings Line 1 Settings (PHONE Port) Field Description DTMF Tx Strict Hold Off Time This parameter is in effect only when DTMF Tx Mode is set to strict, and when DTMF Tx Method is set to out-ofband; i.e. either AVT or SIP-INFO. The value can be set as low as 40 ms. There is no maximum limit.
4 Configuring the Voice Settings Line 1 Settings (PHONE Port) Field Description Symmetric RTP Enable symmetric RTP operation. If enabled, the ATA sends RTP packets to the source address and port of the last received valid inbound RTP packet. If disabled (or before the first RTP packet arrives) the ATA sends RTP to the destination as indicated in the inbound SDP.
4 Configuring the Voice Settings Line 1 Settings (PHONE Port) Gateway Accounts Field Description Gateway1/2/3/4 The first of 4 gateways that can be specified to be used in the to facilitate call routing specification (that overrides the given proxy information). This gateway is represented by gw1 in the . For example, the rule 1408xxxxxxx<:@gw1> can be added to the dial plan such that when the user dials 1408+7digits, the call will be routed to Gateway 1.
4 Configuring the Voice Settings PSTN (LINE Port) Field Description Caller Conn Polarity Polarity after an outbound call is connected: Forward or Reverse. Default setting: Forward. Callee Conn Polarity Polarity after an inbound call is connected: Forward or Reverse. Default setting: Forward PSTN (LINE Port) Use the Voice > PSTN page to configure the settings for calls through the LINE (PSTN) port. To open this page: Click Voice on the menu bar, and then click PSTN in the navigation tree.
4 Configuring the Voice Settings PSTN (LINE Port) Field Description RTP ToS/DiffServ Value ToS/DiffServ field value in UDP IP packets carrying RTP data. Default setting: 0xb8 RTP CoS Value CoS value for RTP data. Valid values are 1 through 7. Default setting: 6 Network Jitter Level Determines how jitter buffer size is adjusted by the ATA device. Jitter buffer size is adjusted dynamically.
4 Configuring the Voice Settings PSTN (LINE Port) Field Description SIP 100REL Enable To enable the support of 100REL SIP extension for reliable transmission of provisional responses (18x) and use of PRACK requests, select yes. Otherwise, select no. Default setting: no EXT SIP Port The external SIP port number. Default setting: 5061 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.
4 Configuring the Voice Settings PSTN (LINE Port) Field Description SIP Debug Option SIP messages are received at or sent from the proxy listen port. This feature controls which SIP messages to log. Choices are as follows: • none—No logging. • 1-line—Logs the start-line only for all messages. • 1-line excl. OPT—Logs the start-line only for all messages except OPTIONS requests/responses. • 1-line excl. NTFY—Logs the start-line only for all messages except NOTIFY requests/responses.
4 Configuring the Voice Settings PSTN (LINE Port) Field Description Restrict Source IP If Lines 1 and 2 use the same SIP Port value and the Restrict Source IP feature is enabled, the proxy IP address for Lines 1 and 2 is treated as an acceptable IP address for both lines. To enable the Restrict Source IP feature, select yes. Otherwise, select no. If configured, the ATA will drop all packets sent to its SIP Ports originated from an untrusted IP address.
4 Configuring the Voice Settings PSTN (LINE Port) Field Description Use Anonymous with RPID When set to yes, use “anonymous” in the SIP message when remote party ID is requested in the SIP message. Default setting: yes Use Local Addr in FROM The IP address of the local address enclosed in the FROM of the SIP message. Default setting: no NAT Settings Field Description NAT Mapping Enable To use externally mapped IP addresses and SIP/RTP ports in SIP messages, select yes. Otherwise, select no.
4 Configuring the Voice Settings PSTN (LINE Port) Proxy and Registration Field Description Proxy SIP proxy server for all outbound requests. Default setting: blank Outbound Proxy SIP Outbound Proxy Server where all outbound requests are sent as the first hop. Default setting: blank Use Outbound Proxy Enable the use of Outbound Proxy. If set to no, the Outbound Proxy parameter and Use OB Proxy in Dialog is ignored.
4 Configuring the Voice Settings PSTN (LINE Port) Field Description Ans Call Without Reg Expires value in sec in a REGISTER request. ATA will periodically renew registration shortly before the current registration expired. This parameter is ignored if the Register parameter is no. Range: 0 – (231 – 1) sec Default setting: yes Use DNS SRV If required by your provider, check this box to use DNS SRV lookup for Proxy and Outbound Proxy.
4 Configuring the Voice Settings PSTN (LINE Port) Field Description Password Password for this line. Use Auth ID To use the authentication ID and password for SIP authentication, select yes. Otherwise, select no to use the user ID and password. Default setting: no Auth ID The Authentication ID for SIP authentication.
4 Configuring the Voice Settings PSTN (LINE Port) Field Description Silence Supp Enable To enable silence suppression so that silent audio frames are not transmitted, select yes. Otherwise, select no. Default setting: no G726-32 Enable To enable the use of the G726 codec at 32 kbps, select yes. Otherwise, select no. Default setting: no Echo Canc Enable To enable the use of the echo canceller, select yes. Otherwise, select no.
4 Configuring the Voice Settings PSTN (LINE Port) Field Description FAX Passthru Method Select the fax passthrough method: None, NSE, or ReINVITE. Default setting: NSE 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.
4 Configuring the Voice Settings PSTN (LINE Port) VoIP-To-PSTN Gateway Setup Field Description VoIP-To-PSTN Gateway Enable Choose yes to enable or choose no to disable the VoIP-ToPSTN Gateway functionality. Default setting: yes VoIP Caller Auth Method The method to authenticate a VoIP Caller to access the PSTN gateway. Choose from none, PIN, or HTTP Digest.
4 Configuring the Voice Settings PSTN (LINE Port) Field Description VoIP Caller ID Pattern A comma-separated list of caller phone number patterns that is used to allow or block access to the PSTN gateway based on the caller ID. If the caller ID does not match a specified pattern, access is rejected, regardless of the authentication method. This comparison is applied before the access list is applied. If this parameter is blank (not specified), all callers are considered for VoIP service.
4 Configuring the Voice Settings PSTN (LINE Port) VoIP Users and Passwords (HTTP Authentication) Field Description VoIP User 1/2/3/4/ 5/6/7/8 Auth ID A user ID that a VoIP Caller can use for authentication by using the HTTP Digest method (in other words, by embedding an Authorization header in the SIP INVITE message sent to the ATA. If the credentials are missing or incorrect, the ATA will challenge the caller with a 401 response).
4 Configuring the Voice Settings PSTN (LINE Port) Field Description PSTN Ring Thru Line 1 To enable ring through to Line 1 based on caller number patterns, choose yes. Otherwise choose no. Note: For more information about PSTN Caller number patterns, see PSTN Caller ID Pattern. Default setting: yes PSTN PIN Max Retry The number of times that a PSTN caller can attempt to enter a PIN number, if the authentication method is set to PIN. Default setting: 3 PSTN CID for VoIP CID Choose yes or no.
4 Configuring the Voice Settings PSTN (LINE Port) Field Description PSTN Caller ID Pattern A comma-separated list of phone number patterns that is used to allow or block access to the VoIP gateway based on the caller ID. If the caller ID does not match a specified pattern, access is rejected, regardless of the authentication method. This comparison is applied before the access list is applied. If this parameter is blank (not specified), all callers are considered for VoIP service.
4 Configuring the Voice Settings PSTN (LINE Port) PSTN Timer Values (sec) Field Description VoIP Answer Delay The number of seconds to wait before auto-answering an inbound VoIP call for the FXO account. The range is 0-255. Default setting: 0 VoIP PIN Digit Timeout After a VoIP caller is prompted for a PIN or enters a digit, the number of seconds to wait for an entry. The range is 0255.
4 Configuring the Voice Settings PSTN (LINE Port) Field Description VoIP DLG Refresh Intvl The interval between (SIP) Dialog refresh messages sent by the ATA to detect if the VoIP call-leg is still up. If this value is set to 0, the VoIP call-leg status will not be checked by the ATA. The refresh message is a SIP ReINVITE, and the VoIP peer must response with a 2xx response. If the VoIP peer does not reply or the response is not greater than 2xx, the ATA will disconnect both call legs automatically.
4 Configuring the Voice Settings PSTN (LINE Port) Field Description Detect Polarity Reversal Choose yes to enable or choose no to disable this feature. If enabled, the ATA will disconnect both call legs when this signal is detected during a gateway call. If it is a PSTN gateway call, the first polarity reversal is ignored and the second one triggers the disconnection. For VoIP gateway call, the first polarity reversal triggers the disconnection.
4 Configuring the Voice Settings PSTN (LINE Port) Field Description Detect Disconnect Tone Choose yes to enable or choose no to disable this feature. If enabled, the ATA will disconnect both call legs when it detects the disconnect tone from the PSTN side during a gateway call. Disconnect tone is specified in the Disconnect Tone parameter, which depends on the region of the PSTN service.
4 Configuring the Voice Settings PSTN (LINE Port) Field Description Disconnect Tone (continued) Disconnect Tone Script values: US—480@-30,620@-30;4(.25/.25/1+2) UK—400@-30,400@-30; 2(3/0/1+2) France—440@-30,440@-30; 2(0.5/0.5/1+2) Germany—440@-30,440@-30; 2(0.5/0.5/1+2) Netherlands—425@-30,425@-30; 2(0.5/0.5/1+2) Sweden—425@-10; 10(0.25/0.25/1) Norway—425@-10; 10(0.5/0.5/1) Italy—425@-30,425@-30; 2(0.2/0.2/1+2) Spain—425@-10; 10(0.2/0.2/1,0.2/0.2/1,0.2/0.6/1) Portugal—425@-10; 10(0.5/0.
4 Configuring the Voice Settings User 1 Field Description PSTN To SPA Gain dB of digital gain (or attenuation if negative) to be applied to the signal sent from the PSTN side to the ATA. The range is -15 to 12. Default setting: 0 Ring Validation Time The minimum signal duration required by the Gateway for recognition as a ring signal. Default setting: 256 ms Ring Indication Delay Choose from {0, 512, 768, 1024, 1280, 1536, 1792} (ms).
4 Configuring the Voice Settings User 1 Field Description Cfwd Busy Dest Forward number for Call Forward Busy Service. Same as Cfwd All Dest. Default setting: blank Cfwd No Ans Dest Forward number for Call Forward No Answer Service. Same as Cfwd All Dest. Default setting: blank Cfwd No Ans Delay Delay in sec before Call Forward No Answer triggers. Same as Cfwd All Dest.
4 Configuring the Voice Settings User 1 Field Description Cfwd Last Dest The destination for the Cfwd Last Caller. Block Last Caller The number of the last caller; this caller is blocked via the Block Last Caller Service. For more information, see Vertical Service Activation Codes. Default setting: blank Accept Last Caller The number of the last caller; this caller is accepted via the Accept Last Caller Service. For more information, see Vertical Service Activation Codes.
4 Configuring the Voice Settings User 1 Field Description Dist Ring Setting Distinctive Ring on or off. Default setting: yes Secure Call Setting If yes, all outbound calls are secure calls by default, without requiring the user to dial a star code first. Default setting: no • If Secure Call Setting is set to yes, all outbound calls are secure. However, a user can disable security for a call by dialing *19 before dialing the target number.
4 Configuring the Voice Settings User 1 Field Description Media Loopback Mode The loopback mode to assume locally when making call to request media loopback. Choices are: Source and Mirror. Default setting: source NOTE If the ATA answers the call, the mode is determined by the caller. Media Loopback Type The loopback type to use when making call to request media loopback operation. Choices are Media and Packet.
4 Configuring the Voice Settings PSTN User Field Description Cfwd Ring Splash Len Duration of ring splash when a call is forwarded (0 – 10.0s) Default setting: 0 Cblk Ring Splash Len Duration of ring splash when a call is blocked (0 – 10.0s) Default setting: 0 VMWI Ring Policy The parameter controls when a ring splash is played when a the VM server sends a SIP NOTIFY message to the ATA indicating the status of the subscriber’s mail box. Three settings are available.
4 Configuring the Voice Settings PSTN User PSTN-To-VoIP Selective Call Forward Settings Field Description Cfwd Sel1-8 Caller Eight PSTN Caller Number Patterns to be blocked for VoIP gateway services or forwarded to a certain VoIP number. If the caller is blocked, the ATA will not auto-answers the call. Cfwd Sel1-8 Dest Eight VoIP destinations to forward a PSTN caller matching the Cfwd Sel x Caller parameter. If this entry is blank, the PSTN caller is blocked for VoIP service.
4 Configuring the Voice Settings DECT Line 1 - DECT Line 10 PSTN Ring Thru Line 1 Ring Settings Field Description Default Ring The default ring to be used to ring through Line 1. Choose from {1,2,3,4,5,6,7,8,Follow Line Cfg}. If Follow Line Cfg is selected, the ring is determined by the distinctive ring settings for Line 1. The ring patterns are configured on the Voice > Regional page. For more information, see Distinctive Ring Patterns, page 85.
4 Configuring the Voice Settings DECT Line 1 - DECT Line 10 Field Description SAS DLG Refresh Intvl If this value is not zero, it is the interval at which the streaming audio server sends out session refresh (SIP reINVITE) messages to determine whether the connection to the caller is still active. If the caller does not respond to the refresh message, the ATA ends this call with a SIP BYE message.
4 Configuring the Voice Settings DECT Line 1 - DECT Line 10 Field Description NAT Keep Alive Enable To send the configured NAT keep alive message periodically, select yes. Otherwise, select no. Default setting: no NAT Keep Alive Msg Enter the keep alive message that should be sent periodically to maintain the current NAT mapping. If the value is $NOTIFY, a NOTIFY message is sent. If the value is $REGISTER, a REGISTER message without contact is sent. Escape sequence of %xx is also accepted.
4 Configuring the Voice Settings DECT Line 1 - DECT Line 10 Field Description Network Jitter Level Determines how jitter buffer size is adjusted by the ATA. Jitter buffer size is adjusted dynamically. The minimum jitter buffer size is 30 milliseconds or (10 milliseconds + current RTP frame size), whichever is larger, for all jitter level settings. However, the starting jitter buffer size value is larger for higher jitter levels.
4 Configuring the Voice Settings DECT Line 1 - DECT Line 10 Field Description Auth ResyncReboot If this feature is enabled, the ATA authenticates the sender when it receives the NOTIFY resync reboot (RFC 2617) message. To use this feature, select yes. Otherwise, select no. Default setting: yes SIP Proxy-Require The SIP proxy can support a specific extension or behavior when it sees this header from the user agent.
4 Configuring the Voice Settings DECT Line 1 - DECT Line 10 Field Description SIP Debug Option SIP messages are received at or sent from the proxy listen port. This feature controls which SIP messages to log. The choices are described below. Default setting: none RTP Log Intvl Cisco SPA232D Administration Guide • none—No logging. • 1-line—Logs the start-line only for all messages. • 1-line excl. OPT—Logs the start-line only for all messages except OPTIONS requests/responses. • 1-line excl.
4 Configuring the Voice Settings DECT Line 1 - DECT Line 10 Field Description Restrict Source IP If Lines 1 and 2 use the same SIP Port value and the Restrict Source IP feature is enabled, the proxy IP address for Lines 1 and 2 is treated as an acceptable IP address for both lines. To enable the Restrict Source IP feature, select yes. Otherwise, select no. If configured, the ATA will drop all packets sent to its SIP Ports originated from an untrusted IP address.
4 Configuring the Voice Settings DECT Line 1 - DECT Line 10 Field Description Auth INVITE When enabled, authorization is required for initial incoming INVITE requests from the SIP proxy. Default setting: no Reply 182 On Call Waiting When enabled, the ATA replies with a SIP182 response to the caller if it is already in a call and the line is off-hook. To use this feature select yes.
4 Configuring the Voice Settings DECT Line 1 - DECT Line 10 Field Description Voice Mail Number The phone number for the voice mail system. Default setting: blank Mailbox ID Enter the ID number of the mailbox for this line. Default setting: blank Proxy and Registration Field Description Proxy SIP proxy server for all outbound requests. Default setting: blank Outbound Proxy SIP Outbound Proxy Server where all outbound requests are sent as the first hop.
4 Configuring the Voice Settings DECT Line 1 - DECT Line 10 Field Description Register Expires Allow answering inbound calls without successful (dynamic) registration by the unit. If proxy responded to REGISTER with a smaller Expires value, the ATA will renew registration based on this smaller value instead of the configured value. If registration failed with an Expires too brief error response, the ATA will retry with the value given in the Min-Expires header in the error response.
4 Configuring the Voice Settings DECT Line 1 - DECT Line 10 Field Description Proxy Redundancy Method The ATA makes an internal list of proxies returned in DNS SRV records. In normal mode this list will contain proxies ranked by weight and priority. If the parameter Based on SRV port is configured, the ATA creates a list in normal mode first, and then inspects the port numbers based on the 1st proxy’s port on the list.
4 Configuring the Voice Settings DECT Line 1 - DECT Line 10 Supplementary Service Subscription The ATA provides native support of a large set of enhanced or supplementary services. All of these services are optional. The parameters listed in the following table are used to enable or disable a specific supplementary service.
4 Configuring the Voice Settings DECT Line 1 - DECT Line 10 Field Description CID–Serv Enable Caller ID Service Default setting: yes CWCID Serv Enable Call Waiting Caller ID Service Default setting: yes Call Return Serv Enable Call Return Service Default setting: yes Call Redial Serv Enable Call Redial Service. Call Back Serv Enable Call Back Service. Three Way Call Serv Enable Three Way Calling Service. Three Way Calling is required for Three Way Conference and Attended Transfer.
4 Configuring the Voice Settings DECT Line 1 - DECT Line 10 Field Description Secure Call Serv Secure Call Service. If this feature is enabled, a user can make a secure call by entering an activation code (*18 by default) before dialing the target number. Then audio traffic in both directions is encrypted for the duration of the call. Default setting: yes For more information about star code settings, see Vertical Service Activation Codes, page 90.
4 Configuring the Voice Settings DECT Line 1 - DECT Line 10 Audio Configuration NOTE 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 G729a resource is already allocated and since only one G.
4 Configuring the Voice Settings DECT Line 1 - DECT Line 10 Field Description Silence Threshold Select the appropriate setting for the threshold: high, medium, or low. Default setting: medium G729a Enable To enable the use of the G729a codec at 8 kbps, select yes. Otherwise, select no. Default setting: no Echo Canc Enable To enable the use of the echo canceller, select yes. Otherwise, select no. Default setting: yes G726-32 Enable To enable the use of the G726 codec at 32 kbps, select yes.
4 Configuring the Voice Settings DECT Line 1 - DECT Line 10 Field Description DTMF Tx Mode DTMF Detection Tx Mode is available for SIP information and AVT. Options are: Strict or Normal. Default setting: Strict for which the following are true: • A DTMF digit requires an extra hold time after detection. • The DTMF level threshold is raised to -20 dBm.
4 Configuring the Voice Settings DECT Line 1 - DECT Line 10 Field Description PSTN Fallback Dial Plan The dial plan used when PSTN fallback is enabled and in use. Default setting: (S0<:@gw0>) Enable IP Dialing Enable or disable IP dialing. If IP dialing is enabled, one can dial [userid@] a.b.c.d[:port], where ‘@’, ‘.’, and ‘:’ are dialed by entering *, user-id must be numeric (like a phone number) and a, b, c, d must be between 0 and 255, and port must be larger than 255.
4 Configuring the Voice Settings DECT Line 1 - DECT Line 10 Field Description Cfwd No Ans Delay Delay in sec before Call Forward No Answer triggers. Same as Cfwd All Dest. Default setting: 20 Supplementary Service Settings Field Description Secure Call Setting If yes, all outbound calls are secure calls by default, without requiring the user to dial a star code first. Default setting: no • If Secure Call Setting is set to yes, all outbound calls are secure.
4 Configuring the Voice Settings DECT Line 1 - DECT Line 10 Field Description Accept Media Loopback Request Controls how to handle incoming requests for loopback operation. Default setting: automatic Media Loopback Mode • never: Never accepts loopback calls; replies 486 to the caller. • automatic: Automatically accepts the call without ringing. • manual: Rings the phone first, and the call must be picked up manually before loopback starts.
4 Configuring the Voice Settings DECT User DECT User Use the Voice > DECT User page to set the user preferences for calls using Cisco SPA302D handsets. To open this page: Click Voice on the menu bar, and then click DECT User in the navigation tree. Enter the settings as described below. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings. General Field Description DECT Enable To enable this handset for service, select yes.
4 Configuring the Voice Settings DECT User Handset 1 Field Description Outgoing Lines A comma-separated list of the index numbers (1~10) for the lines that are available from this handset for an outgoing call. These lines will be listed on the phone screen when the user displays the call options or holds down the green call button. Example: 1,2,8 In this example, a user can select DECT line 1, 2, or 8 for an outbound call.
5 Administration Settings This chapter describes the administrative settings for the ATA. It includes the following sections: • Management • Logging • Diagnostics • Factory Defaults • Firmware Upgrade • Configuration Management • Reboot Management Use the Management pages to manage web access to the configuration utility and to enable protocols for remote configuration and network management.
5 Administration Settings Management Web Access Management Use the Administration > Management > Web Access Management page to configure the settings for access to the administration of the ATA. To open this page: Click Administration in the menu bar, and then click Management > Web Access Management in the navigation tree. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.
5 Administration Settings Management Remote Access Field Description Remote Management Allows access to the configuration utility from a device that is on the WAN side of the ATA. For example, you could connect from another subnet in your office or from your home computer. Click Enabled to enable this feature, or click Disabled to disable it. The default setting is Disabled. The other fields in this section of the page are available only if you enable this feature.
5 Administration Settings Management Field Description Allowed Remote IP Address You can use this feature to limit access to the configuration utility based on the IP address of a device. Choose Any IP Address to allow access from any external IP address. To specify an external IP address or range of IP addresses, select the second radio button and then enter the desired IP address or range. The default setting is Any IP Address.
5 Administration Settings Management Field Description ACS URL The URL for the ACS. The format should be http(s):// xxx.xxx.xxx.xxx:port or xxx.xxx.xxx.xxx:port. The xxx.xxx.xxx.xxx is the domain name or IP address of the ACS server. Both the IP address and the port number are required. ACS Username The username for the ACS. The default username is the Organization Unit Identifier (OUI). This value is required and must match the username configured on the ACS. ACS Password The password for the ACS.
5 Administration Settings Management SNMP Use the Administration > Management > SNMP page to set up Simple Network Management Protocol (SNMP) for the ATA. SNMP is a network protocol that allows network administrators to manage, monitor, and receive notifications of critical events as they occur on the network. The ATA supports SNMPv2 and SNMPv3. It acts as an SNMP agent that replies to SNMP commands from SNMP Network Management Systems. It supports the standard SNMP get, next, and set commands.
5 Administration Settings Management Settings for SNMPv3 Field Description Enabled, Disabled Click Enabled to enable this feature, or click Disabled to disable it. The default setting is Disabled. R/W User Enter the user name for SNMPv3 authentication. The default value is v3rwuser. Auth-Protocol Choose the SNMPv3 authentication protocol from the drop-down list (HMAC-MD5 or HMAC-SHA). Auth-Password Enter the authentication password.
5 Administration Settings Management User List (Password Management) Use the Administration > Management > User List page to manage the two user accounts for the configuration utility. The administrator-level account has the default username admin and password admin. The user-level account has access to modify a limited set of features. This account has the default username cisco and password cisco. For the IVR, no user password is required; the user simply presses # when prompted.
5 Administration Settings Logging Bonjour Use the Administration > Management > Bonjour page to enable or disable Bonjour. Bonjour is a service discovery protocol that locates network devices such as computers and servers on your LAN. It may be required by network management systems that you use. When this feature is enabled, the ATA periodically multicasts Bonjour service records to its entire local network to advertise its existence.
5 Administration Settings Logging Log Module Use the Administration > Log > Log Module page to enable and configure logging. To open this page: Click Administration in the menu bar, and then click Log > Log Module in the navigation tree. NOTE • As a best practice, Cisco recommends that you enable logging only when needed, and disable logging when you finish the investigation. Logging consumes resources and can impact system performance.
5 Administration Settings Logging Field Description Priority The types of events to include in the log. The lowest level of logging is Emergency, which is limited to messages about high impact events. The highest level of logging is Debugging, which includes all message types from Emergency upward. • Emergency: Messages about events, such as an imminent system crash, that make the system unusable. Typically this type of message is broadcast to all users.
5 Administration Settings Logging Field Description Syslog Server Check the box in the heading row to include all services in the log file that is transmitted to the syslog server. Alternatively, check the box for kernel or system to include that service in the log file. Log Setting If logging is enabled on the Administration > Log > Log Module page, the ATA can periodically send the log file to a server or to an email address.
5 Administration Settings Logging E-Mail When logging is enabled, you can send logs to an email address by using SMTP. NOTE Service providers’ requirements vary. Be aware that some providers do not allow SMTP email from a free account. Other providers may require a user to log on to a new mailbox before sending emails. For accurate information, read the support documentation from your provider. In your provider's support or help system, search for information about SMTP server settings.
5 Administration Settings Logging Log Viewer If logging is enabled on the Administration > Log > Log Module page, you can use the Log Viewer page view the logs online and to download the system log file to your computer. You can limit the contents of the log by choosing the types of entries to include and by specifying keywords. NOTE For information about enabling and configuring logging, see Log Module, page 177.
5 Administration Settings Diagnostics Diagnostics The ATA includes two built-in diagnostic tools: • Ping Test • Traceroute Test Ping Test Use the Administration > Diagnostics > Ping Test page to test connectivity between the ATA and a destination. To open this page: Click Administration in the menu bar, and then click Diagnostics > Ping Test in the navigation tree. STEP 1 Enter the IP address or domain name that you want to ping. STEP 2 Enter a packet size in bytes. The range is 32 to 65500 bytes.
5 Administration Settings Factory Defaults The results display up to 30 hops. STEP 3 Click Close to close the results and display the Traceroute Test form. Factory Defaults Use the Administration > Factory Defaults page to reset the ATA to the default configuration. Alternatively, press and hold the RESET button for 20 seconds. All user-changeable non-default settings will be lost. This may include network and service provider data.
5 Administration Settings Configuration Management ! CAUTION Upgrading the firmware may take several minutes. Until the process is complete, DO NOT turn off the power, press the hardware reset button, or click the Back button in your current browser. Configuration Management Use the Administration > Config Management pages to backup and restore the configuration settings for the ATA.
5 Administration Settings Reboot STEP 1 Click Browse to locate the .cfg file on your computer. STEP 2 Click Restore to restore the settings from the selected file. Reboot Use the Administration > Reboot page to power cycle the ATA (if necessary) from the configuration utility. Alternatively, accomplish this task by pressing the RESET button. To open this page: Click Administration in the menu bar, and then click Reboot in the navigation tree. Click the Reboot button to power cycle the ATA.
6 Viewing the Status and Statistics This chapter describes how to view the status and statistics for the ATA. It includes the following sections: • System Information • Interface Information • Internet Status • Port Statistics • DHCP Server Information System Information Use the Status > System Information page to view information about the ATA and its current settings. To open this page: Click Status on the menu bar, and then click System Information in the navigation tree.
6 Viewing the Status and Statistics Interface Information Field Description Host Name The host name of the ATA. Domain Name The domain name of the ATA. Serial Number The serial number of the ATA. Current Time Time that is set on the ATA. Interface Information Use the Status > Interface Information page to view information for the WAN interface (INTERNET port) and the LAN interface (ETHERNET port).
6 Viewing the Status and Statistics Internet Status Field Description Status The status of the port, showing whether the port is connected to a device or disconnected. Clear TX & RX Click this button to reset to 0 the count of TX and RX packets. Internet Status Use the Status > Internet Status page to view information about the WAN interface (INTERNET port). To open this page: Click Status on the menu bar, and then click Internet Status in the navigation tree.
6 Viewing the Status and Statistics Port Statistics Port Statistics Use the Status > Port Statistics page to view information about the port activity on the WAN interface (INTERNET port) and the LAN interface (ETHERNET port). To open this page: Click Status on the menu bar, and then click Port Statistics in the navigation tree. Field Description Input (pkts) The number of packets received by the port. Output (pkts) The number of packets transmitted by the port.
6 Viewing the Status and Statistics DHCP Server Information DHCP Server Information Use the Status > DHCP Server Information page to view information about the DHCP server and clients. To open this page: Click Status on the menu bar, and then click DHCP Server Information in the navigation tree. DHCP Pool Information Field Description Client Name The host name of the DHCP client. IP Address The IP address leased to the client. MAC Address The MAC address of the DHCP client.
6 Viewing the Status and Statistics DHCP Server Information Field Description Client Lease Time The maximum amount of time, in minutes, that a client can lease a dynamically assigned IP address. Static DNS The IP addresses of up to three DNS servers to be used by DHCP clients. Option 66 The setting for Option 66, which provides provisioning server address information to hosts requesting this option. The ATA may be set to None (internal), Remote TFTP Server, or Manual TFTP Server.
A Frequently Asked Questions Q. I cannot connect to the Internet through the ATA. STEP 1 Make sure that the ATA is powered on. The Power/Sys LED should be solid green and not flashing. If the Power LED is flashing, then power off all of your network devices, including the modem, the ATA, and the connected devices. Wait for 30 seconds. Then power on each device in the following order: • Cable or DSL modem • ATA • Connected Devices STEP 2 Check the cable connections.
A Frequently Asked Questions Q. There is no dial tone, and the Phone 1 or 2 LED is not solid green. STEP 1 Make sure the telephone is connected to the appropriate port, PHONE 1 or 2. STEP 2 Disconnect the RJ-11 telephone cable from the PHONE port, and then reconnect it. STEP 3 Make sure your telephone is set to its tone setting (not pulse). STEP 4 Make sure your network has an active Internet connection. Try to access the Internet, and check to see if the ATA WAN LED is flashing green.
Frequently Asked Questions A Q. When I open a web browser, I am prompted for a username and password. How can I bypass this prompt? Launch the web browser and perform the following steps (these steps are specific to Internet Explorer but are similar for other browsers): STEP 1 Select Tools > Internet Options. STEP 2 Click the Connections tab. STEP 3 Select Never dial a connection. STEP 4 Click OK. Q. The DSL telephone line does not fit into the ATA WAN (Internet) port.
B Using the IVR for Administration An IVR system is available to help you to configure and manage your ATA. Use a telephone keypad to select options and to make your entries. To access the IVR menu: STEP 1 Connect an analog phone to a PHONE port of the ATA. STEP 2 Press the star key four times: **** STEP 3 When challenged for a password, log in as an administrator by using the default administrator's password of 1234# or log in as the PHONE port's user by pressing #.
B Using the IVR for Administration • If the menu is inactive for more than one minute, the IVR times out. You will need to re-enter the IVR menu by pressing the star key four times: ****. Your settings take effect after you hang up the telephone or exit the IVR. The ATA may reboot at this time. • To enter the decimal points in an IP address, press the * (star) key. For example, to enter the IP address 191.168.1.
B Using the IVR for Administration IVR Action Menu Option Choices and Instructions Set Network Mask 121 To enter the value, press numbers on the telephone key pad. Press the * (star) key to enter a decimal point. Note: This option is available only after you choose Static IP as the Internet Connection Type, through option 101. Check Gateway IP Address 130 Set Gateway IP Address 131 To enter the value, press numbers on the telephone key pad. Press the * (star) key to enter a decimal point.
B Using the IVR for Administration IVR Action Menu Option Choices and Instructions Set Line 1 SIP Transport 1911 0: UDP 1: TCP 2: TLS Check Line 2 SIP Transport 1920 Set Line 2 SIP Transport 1921 0: UDP 1: TCP 2: TLS Exit IVR 3948 (Spells EXIT on the phone keypad) Allow or prevent WAN access to the administration web server 7932 Factory Reset of Unit 73738 WARNING: All nondefault settings will be lost. This includes network and service provider data.
B Using the IVR for Administration IVR Action Menu Option Choices and Instructions User Factory Reset of Unit 877778 When prompted, press 1 to confirm, or press * (star) to cancel. After you hear “Option successful,” hang up the phone. The ATA reboots. WARNING: All userchangeable non-default settings will be lost. This may include network and service provider data.
C Advanced Options for Voice Services This appendix provides additional information about configuring advanced options for voice services. STEP 1 • Optimizing Fax Completion Rates • VoIP-to-PSTN and PSTN-to-VoIP Calling • Call Scenarios • Configuring Dial Plans Optimizing Fax Completion Rates Issues can occur with fax transmissions over IP networks, even with the T.38 standard, which is supported by the ATA. You can adjust several settings on your ATA to optimize your fax completion rates.
Advanced Options for Voice Services Optimizing Fax Completion Rates • C Three Way Call Serv: no STEP 5 In the Audio Configuration section, enter the following settings to support T.38 fax: • Preferred Codec: G.711u (USA) or G.711a (rest of the world) • Use pref. codec only: yes • Silence Supp Enable: no • Echo Canc Enable: no • FAX Passthru Method: ReINVITE STEP 6 Click Submit to save your settings or click Cancel to abandon the unsaved settings.
Advanced Options for Voice Services Optimizing Fax Completion Rates C STEP 3 Determine the success rate. STEP 4 Monitor the network and record the statistics for jitter, loss, and delay. STEP 5 If faxes fail consistently, capture a copy of the configuration as described below. You can then send this file to Technical Support. a. In your web browser, enter the path for the configuration file: http:///admin/config.xml&xuser= &xpassword= b.
C Advanced Options for Voice Services VoIP-to-PSTN and PSTN-to-VoIP Calling VoIP-to-PSTN and PSTN-to-VoIP Calling The ATA allows calls to be made by using SIP-based Voice-over-IP (VoIP) services and traditional telephone Public Switched Telephone Network (PSTN) services. Calls can be placed and received by using an analog phone or fax machine and Cisco SPA302D Mobility Enhanced Cordless Telephone Handsets.
Advanced Options for Voice Services VoIP-to-PSTN and PSTN-to-VoIP Calling C The credentials are computed based on the corresponding password using Message Digest 5 (MD5). The parameter must match one of the VoIP accounts stored on the ATA device. You can configure these settings on the PSTN (LINE Port) page. Two-Stage Dialing In two-stage dialing, the LINE port goes off-hook but does not automatically dial any digits after accepting the call.
Advanced Options for Voice Services VoIP-to-PSTN and PSTN-to-VoIP Calling C PSTN callers can be authenticated by one of the following methods: • No authentication—All callers are accepted for service. • PIN—Caller is prompted to enter a PIN right after the call is answered. NOTE When the source address of the INVITE is 127.0.0.1, authentication is automatically disabled because this is a call by the local user. This applies to both one-stage and two-stage dialing.
Advanced Options for Voice Services VoIP-to-PSTN and PSTN-to-VoIP Calling C Terminating Gateway Calls There are two call legs in a PSTN gateway call: the PSTN call leg and the VoIP call leg. A gateway call is terminated when either call leg is ended. When the call terminates, the LINE port goes on-hook so the PSTN line is available for use.
C Advanced Options for Voice Services VoIP-to-PSTN and PSTN-to-VoIP Calling • pwd: Password used for authentication with the given gateway • nat: Enable or disable NAT mapping when calling the gateway The following table lists some examples. Example Description <9,:>xx.<:@gw1 Dial 9 to start outside dial tone, followed by one or more digits, and route the call to Gateway 1. [93]11<:@gw0> Route 911 and 311 calls to the local PSTN gateway <8,:1408>xxxxxxx<:@pstn. cisco.
Advanced Options for Voice Services VoIP-to-PSTN and PSTN-to-VoIP Calling C Sharing One VoIP Account Between the PHONE and LINE Ports Both the PHONE port (Line 1) and LINE port (PSTN) can receive incoming calls for a single VoIP account. Consider the following points: • If the service provider allows multiple registration contacts and simultaneous ringing, both lines can register periodically with the service provider. In this case, both lines receive inbound calls to this VoIP account.
C Advanced Options for Voice Services VoIP-to-PSTN and PSTN-to-VoIP Calling PSTN Call to Ring Line 1 This feature allows a PSTN caller to ring Line 1. When the PSTN line rings, the PSTN Line makes a local VoIP call to Line 1. If Line 1 is busy, it stops. After a given number of rings, the VoIP gateway picks up the call. Symmetric RTP The Symmetric RTP parameter is used to send audio RTP to the source IP and port of the inbound RTP packets. This facilitates NAT traversal.
Advanced Options for Voice Services Call Scenarios C Call Scenarios This section describes some typical scenarios where the ATA device can be applied. Some terms are introduced in the first few sections and reused in later sections.
Advanced Options for Voice Services Call Scenarios C Using PIN Authentication This scenario assumes that the PSTN Line has a different VoIP account than the Line 1 account. The VoIP caller calls the FXO number, which auto-answers after VoIP Answer Delay. The ATA device then prompts the VoIP caller for a PIN. When a valid PIN is entered, the ATA plays the Outside Dial Tone and prompts the caller to dial the PSTN number. The number dialed is processed by the dial plan corresponding to the VoIP caller.
Advanced Options for Voice Services Call Scenarios C NOTE HTTP Digest Authentication is one way to perform one-stage dialing of a VoIP-To- PSTN call. The other way is with no authentication require. However, if the target number is not specified in the Request-URI or the number matches the account user-id of the PSTN Line, the call reverts to two-stage dialing.
Advanced Options for Voice Services Configuring Dial Plans C Forward-All to the PSTN gateway In this scenario, Line 1 is configured with Cfwd All Dest parameter to the PSTN gateway.This scenario is the same the previous case, except the FXO picks up the Line 1 call immediately. If the PSTN Line is busy at the moment of the call, the PSTN Line does not pick up the call, the call forward rule is ignored, and Line 1 continues to ring.
C Advanced Options for Voice Services Configuring Dial Plans • Interdigit Short Timer (Complete Entry Timer) • Resetting the Control Timers Digit Sequences A dial plan contains a series of digit sequences, separated by the pipe character: | The entire collection of sequences is enclosed within parentheses. Each digit sequence within the dial plan includes a series of elements, which are individually matched to the keys that the user presses.
C Advanced Options for Voice Services Configuring Dial Plans Digit Sequence Function 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.
Advanced Options for Voice Services Configuring Dial Plans C 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.
Advanced Options for Voice Services Configuring Dial Plans C 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.
C Advanced Options for Voice Services Configuring Dial Plans 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.
C Advanced Options for Voice Services Configuring Dial Plans Terminating Event Processing The user presses the # key. • If the sequence is complete and is allowed by the dial plan, the number is accepted and is transmitted according to the dial plan. • If the sequence is incomplete or is blocked by the dial plan, the number is rejected. Dial Plan Timer (Off-Hook Timer) You can think of the Dial Plan Timer as “the off-hook timer.” This timer starts counting when the phone goes off hook.
Advanced Options for Voice Services Configuring Dial Plans C allowing 0 or more digits. This setting will produce unwanted results especially if you are deploying timers. • Create a hotline for all sequences on the System Dial Plan (P9<:23> | (9,8<:1408>[2-9]xxxxxx | 9,8,1[2-9]xxxxxxxxx | 9,8,011xx. | 9,8,xx.|[1-8]xx) P9<:23> After taking the phone off hook, a user has 9 seconds to begin dialing. If no digits are pressed within 9 seconds, the call is transmitted automatically to extension 23.
Advanced Options for Voice Services Configuring Dial Plans C 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.
Advanced Options for Voice Services Configuring Dial Plans C STEP 1 Log in to the configuration utility. If prompted, enter the administrative logon provided by the Service Provider. The default username and password are both admin. STEP 2 Under the Voice menu, click Regional. STEP 3 In the Control Timer Values section, enter the desired values in the Interdigit Long Timer field and the Interdigit Short Timer field. See the definitions at the beginning of this section.
D Where to Go From Here Cisco provides a wide range of resources to help you and your customer obtain the full benefits of the ATA. Support Cisco Small Business Support Community www.cisco.com/go/smallbizsupport Online Technical Support and Documentation (Login Required) www.cisco.com/support Cisco Small Business Support and Resources www.cisco.com/go/smallbizhelp Downloads and Documentation Firmware www.cisco.com/go/software Cisco Small Business Voice Gateways Documentation www.cisco.
