Cisco Small Business SPA300 Series, SPA500 Series, and WIP310 IP Phone Administration Guide

Connection Type

Indicates the type of internet connection for the phone:

• DHCP
• Static IP
• PPPoE (only applicable to Cisco SPA525G or Cisco SPA525G2)

Current IP

Displays the current IP address assigned to the IP phone.

Host Name

Displays the current host name assigned to the SPA9000 (defaults to SipuraSPA).

Domain

Displays the network domain name of the SPA9000.

Current Netmask

Displays the network mask assigned to the SPA9000.

Current Gateway

Displays the default router assigned to the SPA9000.

Primary DNS

Displays the primary DNS server assigned to the SPA9000.

Secondary DNS

Displays the secondary DNS server assigned to the SPA9000.

NTP Enable

Shows if Network Time Protocol is enabled.

Primary NTP Server

IP Address of the primary NTP server.

Secondary NTP Server

IP Address of the secondary NTP server.

TFTP Server

Address of the TFTP server for provisioning.

Bluetooth Enabled

Shows if Bluetooth is enabled.

Bluetooth Firmware Version

Displays the Bluetooth firmware version.

Bluetooth Connected

Shows if a Bluetooth device is connected to the phone.

Bluetooth MAC

Shows the hardware address of the Bluetooth device.

Connected Device ID

Shows the name of the connected Bluetooth device.

Wireless Enabled

Shows if Wireless-G is enabled on the phone.

Wireless Connected

Shows if the phone is connected to the wireless network.

Wireless MAC

Shows the hardware address of the Wireless-G controller.

SSID

Shows the SSID, or name of the wireless router to which the phone is connected.

Standard Channel

Shows the wireless channel being used in the wireless connection.

Security Mode

Shows if wireless security is configured on the phone (yes or no).

Upgrade

An upgrade operation caused a reboot (regardless whether the upgrade completed or failed).

Provisioning

Changes made to parameter values by using the phone LCD or Web GUI, or a resync caused a reboot.

SIP Triggered

A SIP request caused a reboot.

RC

A remote customization caused a reboot.

User Triggered

The user manually triggered a warm reboot.

Software Req

A remote server triggered a warm reboot.

System n

System events (for example, running out of resources) triggered a warm reboot.

IP Changed

The phone IP address was changed triggering a warm reboot.

TFTP Server

Address of the TFTP server for provisioning.

Call Manager

IP address of the Unified Communications server.

Directories URL

Populated by the Unified Communications Server; points to the directory application server.

Services URL

Populated by the Unified Communications Server; points to the Cisco XML application server.

Authentication URL

Populated by the Unified Communications Server; points to the authentication server.

DHCP Address Released

Populated by the Unified Communications Server; indicates if the DHCP address has been released.

VPN Connected

Indicates if the phone is connected to a VPN.

Client Address

IP address given to the phone from the VPN server.

Client Netmask

Netmask given to the phone from the VPN server.

Bytes Sent

Size of data sent from the phone.

Bytes Recv

Size of data received by the phone.

Product Name

Model number of the IP phone.

Serial Number

Serial number of the IP phone.

Software Version

Version number of the IP phone software.

Hardware Version

Version number of the IP phone hardware.

MAC Address

Hardware address of the IP phone.

Client Certificate

Status of the client certificate, which authenticates the IP phone for use in the ITSP network. This field indicates if the client certificate is properly installed in the IP phone.

Customization

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

Licenses

Indicates any additional licenses that you have installed in the IP phone.

Current Time

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

Elapsed Time

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

Broadcast Pkts Sent

Total number of broadcast packets sent.

Broadcast Bytes Sent

Total number of broadcast packets received.

Broadcast Pkts Recv

Total number of broadcast bytes sent.

Broadcast Bytes Recv

Total number of broadcast bytes received and processed.

Broadcast Pkts Dropped

Total number of broadcast packets received but not processed. Most codecs can handle up to 5% random packet drops as long as the packets are random and not in groups of two or more. Concurrent packet drops result in voice quality issues.

Broadcast Bytes Dropped

Total number of broadcast bytes received but not processed.

RTP Packets Sent

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

RTP Bytes Sent

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

RTP Packets Recv

Total number of RTP bytes sent.

RTP Bytes Recv

Total number of RTP bytes received.

SIP Messages Sent

Total number of SIP messages sent (including retransmissions).

SIP Bytes Sent

Total number of SIP messages received (including retransmissions).

SIP Messages Recv

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

SIP Bytes Recv

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

External IP

External IP address used for NAT mapping.

Operational VLAN ID

ID of the VLAN currently in use if applicable.

Note Not applicable to Cisco WIP310.

SW Port (Cisco SPA300 Series and Cisco SPA500 Series)

Displays the type of Ethernet connection from the IP phone to the switch.

PC Port (Cisco SPA303 and Cisco SPA500 Series)

Indicates whether the link from the IP phone to a device plugged into the PC port on the phone is up or down.

Registration State

Shows “Registered” if the phone is registered, “Not Registered” if the phone is not registered to the ITSP.

Last Registration At

Last date and time the line was registered.

Next Registration In

Number of seconds before the next registration renewal.

Message Waiting

Indicates whether the phone user has a new voice mail waiting: Yes or No. This is updated when voice mail notification is received.

Mapped SIP Port

Port number of the SIP port mapped by NAT.

Hoteling State

Indicates the status of the phone with the guest account.

This parameter is of 6 types:

• Disabled: When the parameter, <Enable Broadsoft Hoteling> is set to no.
• Unsubscribed: When the parameter, <Enable Broadsoft Hoteling> is set to yes, and the Hoteling State is not subscribed to server.
• Associating xxx: When the phone is trying for guest login, where xxx stands for the guest User ID.
• Associated xxx: When the guest login is successful on the phone..
• Disassociating: When the phone is trying to disassociate with the guest account.
• Disassociated: When the phone has disassociated with the guest account.

Call State

Status of the call.

Tone

Type of tone used by the call.

Encoder

Codec used for encoding.

Decoder

Codec used for decoding.

Type

Direction of the call.

Remote Hold

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

Callback

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

Peer Name

Name of the internal phone.

Peer Phone

Phone number of the internal phone.

Duration

Duration of the call.

Packets Sent

Number of packets sent.

Packets Recv

Number of packets received.

Bytes Sent

Number of bytes sent.

Bytes Recv

Number of bytes received.

Mapped RTP Port

The port mapped for Real Time Protocol traffic for the call.

Media Loopback

If the call is a loopback call, displays the loopback mode (source or mirror) and type (media or packet). If the call is not loopback, the field appears blank.

Decode Latency

Number of milliseconds for decoder latency.

Jitter

Number of milliseconds for receiver jitter.

Round Trip Delay

Number of milliseconds for delay in the RTP-to-RTP interface round trip.

End System Delay

Number of milliseconds for delay in the internal round trip within the reporting endpoint.

Packets Lost

Number of packets lost.

Packet Error

Number of invalid packets received.

Loss Rate

The fraction of RTP data packets from the source lost since the beginning of reception. Defined in RFC-3611—RTP Control Protocol Extended Reports (RTCP XR).

Discard Rate

The fraction of RTP data packets from the source that have been discarded since the beginning of reception, due to late or early arrival, under-run or overflow at the receiving jitter buffer. Defined in RFC-3611—RTP Control Protocol Extended Reports (RTCP XR).

Burst Duration

The mean duration, expressed in milliseconds, of the burst periods that have occurred since the beginning of reception. Defined in RFC-3611—RTP Control Protocol Extended Reports (RTCP XR).

Gap Duration

The mean duration, expressed in milliseconds, of the gap periods that have occurred since the beginning of reception. Defined in RFC-3611—RTP Control Protocol Extended Reports (RTCP XR).

R Factor

Voice quality metric describing the segment of the call that is carried over this RTP session. Defined in RFC-3611—RTP Control Protocol Extended Reports (RTCP XR).

MOS-LQ/MOS Listening

The estimated mean opinion score for listening quality (MOS-LQ) is a voice quality metric on a scale from 1 to 5, in which 5 represents excellent and 1 represents unacceptable. Defined in RFC-3611—RTP Control Protocol Extended Reports (RTCP XR).

MOS-CQ/MOS Conversational

The estimated mean opinion score for conversational quality (MOS-CQ) is defined as including the effects of delay and other effects that would affect conversational quality. Defined in RFC-3611—RTP Control Protocol Extended Reports (RTCP XR).

Status

Indicates whether the phone is downloading a ring tone (and from where) or if it is idle.

Ring Tone 1

Information about the user downloaded ring tone 1: name, size, and time-stamp of the tone.

Ring Tone 2

Information about the user downloaded ring tone 2: name, size, and time-stamp of the tone.

Custom CA Provisioning Status

Indicates whether provisioning using a custom CA succeeded or failed:

• Last provisioning succeeded on mm/dd/yyyy HH:MM:SS; or
• Last provisioning failed on mm/dd/yyyy HH:MM:SS

Custom CA Info

Displays information about the custom CA:

• Installed—Displays the “CN Value,” where “CN Value” is the value of the CN parameter for the Subject field in the first certificate.
• Not Installed—Displays if no custom CA certificate is installed.

Restricted Access Domains
(SIP)

This feature is used when implementing software customization.

Enable Web Server

Enable/disable web server of the IP phone.

Defaults to yes.

Web Server Port

Port number of the phone web user interface.

Defaults to 80.

SPA525-http-write (Cisco SPA525G or Cisco SPA525G2 SPCP only)

Allow Cisco Configuration Assistant (CCA) or other application to write XML file parameters directly to the phone using HTTP.

Choose yes to allow this feature, or no to disable this feature.

Enable Web Admin Access

Lets you enable or disable local access to the phone web user interface. Select yes or no from the drop-down menu.

Defaults to yes.

Admin Passwd

Password for the administrator.

Defaults to no password.

User Password

Password for the user.

Defaults to no password.

User Web Password

Eliminates the user requirement for daily password entry on the LCD GUI to use the basic functions. Basic functions include CFwd, Directory, Redial, and Corporate Directory.

The options are No and Yes:

• If the User Password field is not empty, using this field restricts user access through the web interface and LCD GUI.
• If the User Password field is empty, no password is required to access the LCD GUI. However, the user must enter a password to access the web interface. This is the User Web Password parameter that was set earlier.

Default value is empty.

Note This field is supported in Firmware Release 7.6.2 and later.

SPA525-protocol (Cisco SPA525G or Cisco SPA525G2 only)

Allows you to choose the type of protocol for the phone:

• SIP—Session Initiation Protocol. Choose if the phone is used with a SIP call control system, such as the Cisco SPA9000 or a SIP call control system from another provider such as BroadSoft or Asterisk.
• SPCP—Smart Phone Control Protocol. Choose if the phone is used with a Cisco Unified Communications Series server, such as the Cisco Unified Communications 500 Series for Small Business.

SPA525-auto-detect-sccp (Cisco SPA525G or Cisco SPA525G2 only)

Choose if the phone should automatically detect the type of protocol used on the network to which it is connected. If set to yes, the phone automatically discovers if it is connected to a call control system using SPCP.

SPA525-readonly
(Cisco SPA525G or Cisco SPA525G2 only)

If set to yes, the Signaling Protocol and Auto Detect SCCP Settings on the phone are read only. If set to no, the above settings on the phone can be changed by the end user.

Phone-UI-user-mode

Allows you to restrict the menus and options that phone users see when they use the phone interface. Choose yes to enable this parameter and restrict access. The default is no.

Specific parameters are then designated as “na” or “ro” using provisioning files. Parameters designated as “na” will not appear on the phone interface. Parameters designated as “ro” will not be editable by the user.

Note For the SPA525 phones, both the parameters, na and ro, work in a similar fashion.

WiFi User Access Mode

(Cisco SPA525G or Cisco SPA525G2 only)

Setting the parameter to yes allows a user to view and configure Wi-Fi settings, even if the user cannot change other phone parameters because the Phone-UI-user-mode parameter is set to yes.

If set to no, users cannot view and configure WiFi settings if the Phone-UI-user-mode parameter is set to yes.

PoE Power Required

Specifies the PoE power setting: Normal (default) or Maximum.

When one or more attendant consoles are attached to the phone, use Maximum to advertise to a PoE switch that the phone will consume up to 12 watts.

When no attendant consoles are attached, use Minimum to advertise a required power budget of 6.5 watts.

This parameter is not applicable to Cisco SPA301 or SPA303 as they do not have Cisco Attendant Console support.

Connection Type

Choose the type of internet connection:

• DHCP
• Static IP
• PPPoE (not applicable to Cisco WIP310)

Use Backup IP

(Cisco SPA 30X and Cisco SPA5XX only)

When this field is set to yes, the IP phone does not renew its IP address when the DHCP server is not reachable during the phone’s startup sequence, but uses the last-known IP address. If set to no, the phone keeps retrying the DHCP server to obtain a new IP address. This field is set to yes by default.

Static IP

If static IP was chosen as the type of internet connection, displays the static IP address assigned to the phone.

Netmask

If static IP was chosen as the type

Gateway

Default router IP address. Blank if DHCP assigned.

LAN MTU

(Cisco SPA525G or Cisco SPA525G2 SIP mode only)

LAN Maximum Transmission Unit size. Default value: 1500.

Ethernet MTU (Cisco SPA525G or Cisco SPA525G2 SPCP mode only)

Ethernet Maximum Transmission Unit size. Default value: 1500.

Duplex Mode

(Cisco SPA525G or Cisco SPA525G2 SPCP mode only)

Duplex Mode—Choose one of the following to configure the speed/duplex for the phone Ethernet ports:

• Auto
• 10MBps/Duplex
• 10MBps/Half
• 100Mbps/Duplex
• 100MBps/Half

PPPoE Login Name

Specifies the account name assigned by the ISP for connecting on a Point-to-Point Protocol over Ethernet (PPPoE) link.

PPPoE Login Password

Specifies the password assigned by the ISP for connecting on a Point-to-Point Protocol over Ethernet (PPPoE) link.

PPPoE Service Name

Specifies the service name assigned by the ISP for connecting on a Point-to-Point Protocol over Ethernet (PPPoE) link.

Host Name

The host name of the SPA9000.

Domain

The network domain of the SPA9000.

Primary DNS

DNS server used by SPA9000 in addition to DHCP supplied DNS servers if DHCP is enabled; when DHCP is disabled, this is the primary DNS server.

Defaults to 0.0.0.0.

Secondary DNS

DNS server used by SPA9000 in addition to DHCP supplied DNS servers if DHCP is enabled; when DHCP is disabled, this is the secondary DNS server.

Defaults to 0.0.0.0.

DNS Server Order

Specifies the method for selecting the DNS server. The options are Manual, Manual/DHCP, and DHCP/Manual.

DNS Query Mode

Do parallel or sequential DNS Query. With parallel DNS query mode, the SPA9000 sends the same request to all the DNS servers at the same time when doing a DNS lookup, the first incoming reply is accepted by the SPA9000.

Defaults to parallel.

DNS Query TTL Ignore

(Not applicable to Cisco SPA525G or Cisco SPA525G2.)

If enabled, the phone continues to use the previous cached DNS result if the DNS server does not respond when the phone tries to renew its DNS query.

When disabled, the phone uses the previous TTL value and clears the DNS query result cache.

Syslog Server

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

Debug Server

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

Debug Level

The debug level from 0-3. The higher the level, the more debug information is generated. Zero (0) means no debug information is generated. To log SIP messages, you must set the Debug Level to at least 2.

Defaults to 0.

Layer 2 Logging

Used for IP phone network layer debugging purposes. Do not use except when advised to do so by Cisco technical support, as this may impact system performance. Set to no by default.

Primary NTP Server

IP address or name of primary NTP server.

Secondary NTP Server

IP address or name of secondary NTP server.

Enable Bonjour (Cisco SPA525G or Cisco SPA525G2 only)

Enable Bonjour networking that is used by Office Manager and Cisco Configuration Assistant to discover the Cisco IP phones. Choose yes to enable or no to disable.

Enable SSLv3

Choose Yes to enable SSLv3. Choose no to disable.

Defaults to NO.

Enable VLAN

Choose Yes to enable VLAN. Choose no to disable.

VLAN ID

If you use a VLAN without CDP (VLAN enabled and CDP disabled), enter a VLAN ID for the IP phone. Note that only voice packets are tagged with the VLAN ID. Do not use 1 for the VLAN ID.

Enable PC Port VLAN Tagging

Enables VLAN and priority tagging on the phone data port (802.1p/q). This feature facilities tagging of the VLAN ID (802.1Q) and priority bits (802.1p) of the traffic coming from the PC port of the IP phone.

Defaults to No. Choose Yes to enable the tagging algorithm.

PC Port VLAN Highest Priority

0-7 (default 0). The priority applied to all frames, tagged and untagged. The phone modifies the frame priority only if the incoming frame priority is higher than this value.

PC Port VLAN ID

0-4095 (default 0). Value of the VLAN ID. The phone tags all the untagged frames coming from the PC (it will not tag frames with an existing tag).

Enable CDP

Enable CDP only if you are using a switch that has Cisco Discovery Protocol. CDP is negotiation based and determines which VLAN the IP phone resides in.

Enable LLDP-MED

Choose yes to enable LLDP-MED for the phone to advertise itself to devices that use that discovery protocol.

When the LLDP-MED feature is enabled, after the phone has initialized and Layer 2 connectivity is established, the phone sends out LLDP-MED PDU frames. If the phone receives no acknowledgment, the manually configured VLAN or default VLAN will be used if applicable. If the CDP is used concurrently, the waiting period of 6 seconds is used. The waiting period will increase the overall startup time for the phone.

Network Startup Delay

Setting this value causes a delay for the switch to get to the forwarding state before the phone will send out the first LLDP-MED packet. The default delay is 3 seconds. For configuration of some switches, you might need to increase this value to a higher value for LLDP-MED to work. Configuring a delay can be important for networks that use Spanning Tree Protocol.

SPA525-wifi-on

Set to yes to enable Wireless-G service.

Enable BT

Set to yes to enable support for Bluetooth devices.

VPN Server

The IP address of the VPN server to which the phone connects.

VPN User Name

Username configured on the VPN server for the phone.

VPN Password

Password associated with the username configured on the VPN for the phone.

VPN Tunnel Group

(Optional) The tunnel group, if required by the VPN server.

Connect on Bootup

If the phone should attempt to connect to the VPN each time it is powered on. Choose yes to have the phone try to automatically connect, or no to keep the default behavior.

Asset ID

Provides the ability to enter an asset ID for inventory management when using LLDP-MED. The default value for Asset ID is empty. Enter a string of less than 32 characters if you are using this field.

The Asset ID can be provisioned only by using the web management interface or remote provisioning. The Asset ID is not displayed on the phone screen.

Changing the Asset ID field causes the phone to reboot.

Max Forward

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

Defaults to 70.

Max Redirection

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

Defaults to 5.

Max Auth

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

Defaults to 2.

SIP User Agent Name

Used in outbound REGISTER requests.

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

SIP Server Name

Server header used in responses to inbound responses.

Defaults to $VERSION.

SIP Reg User Agent Name

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

Defaults to blank.

SIP Accept Language

Accept-Language header used. To access, click the SIP tab, and fill in the SIP Accept Language field.

There is no default (this indicates SPA9000 does not include this header). If empty, the header is not included.

DTMF Relay MIME Type

MIME Type used in a SIP INFO message to signal a DTMF event. This field must match that of the Service Provider.

Defaults to application/dtmf-relay.

Hook Flash MIME Type

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

The default is application/hook-flash.

Remove Last Reg

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

Defaults to no.

Use Compact Header

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

Default: no

Escape Display Name

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

Defaults to yes.

Escape Special Character

Normally, the IP phone sends a “%23” (escape) as part of the message when the special character # is included in the SIP INVITE message. This can cause problem s with some telephony servers that need to receive the # character. When this parameter is set to yes, the phone sends a “%23” (escape) as part of the message when the # is included. When set to no, the # is sent directly and the escape (%23) is not used.

Defaults to yes.

SIP-B Enable

Enables SIP for Business (supports Sylantro call flows) call features.

Talk Package

Enables support for the BroadSoft Talk Package, which enables a user to answer or resume a call by clicking a button in an external application.

Hold Package

Enables support for the BroadSoft Hold Package, which enables a user to place a call on hold by clicking a button in an external application.

Conference Package

Enables support for the BroadSoft Conference Package, which enables a user to start a conference by clicking a button in an external application.

Notify Conference

If enabled, the unit will send out a NOTIFY with event=conference when starting a conference.

RFC 2543 Call Hold

If set to yes, unit will include c=0.0.0.0 syntax in SDP when sending a SIP re-INVITE to the peer to hold the call. If set to no, unit will not include the c=0.0.0.0 syntax in the SDP. The unit will always include a=sendonly syntax in the SDP in either case.

Defaults to yes.

Random REG CID On Reboot

If set to yes, the Cisco IP phone uses a different random call-ID for registration after the next software reboot. If set to no, the Cisco IP phone tries to use the same call-ID for registration after the next software reboot. The Cisco IP phone always uses a new random Call-ID for registration after a power-cycle, regardless of this setting.

Defaults to no.

Mark All AVT packets

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

Defaults to yes.

SIP TCP Port Min

Specifies the lowest TCP port number that can be used for SIP sessions. Defaults to 5060.

SIP TCP Port Max

Specifies the highest TCP port number that can be used for SIP sessions. Defaults to 5080.

CTI Enable

The CTI interface allows a third-party application to control and monitor the state of a phone that has registered with the Cisco SPA9000. With this interface, an application can control a phone to initiate an outgoing call or answer an incoming call with a mouse click from a PC.

Caller ID Header

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

SRTP Method

Selects the method to use for SRTP. Two choices are available:

• x-sipura—legacy SRPT method
• s-descriptor—new method compliant with RFC-3711 and RFC-4568

The default value is "x-sipura.”

Note Not applicable to Cisco WIP310.

Hold Target Before REFER

Controls whether to hold call leg with transfer target before sending REFER to the transferee when initiating a fully-attended call transfer (where the transfer target has answered). Default value is "no,” where the call leg is not held.

Note Not applicable to Cisco WIP310.

Dialog SDP Enable

When enabled and the Notify message body is too big causing fragmentation, the Notify message xml dialog is simplified; Session Description Protocol (SDP) is not included in the dialog xml content.

Keep Referee When REFER Failed

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

You can also configure this parameter in the configuration file:

Display Diversion Info

Set to yes to cause the IP phone to parse and display the Diversion header in the incoming INVITE messages. The phone displays the information from the first Diversion header; if there are multiple Diversion headers, the others are ignored. The Diversion header follows the definition from RFC 5806. For example, for this diversion header in the INVITE messages:

The screen should display:

The reason field in the header is ignored. If the header field is not properly formed, Diversion header (all) will be ignored. If the privacy field is not set or is set to off, only the URI is displayed due to screen size limitation. If the privacy field is set to full, “anonymous” is displayed. If the privacy field is set to name, only the display name is displayed.

Display Anonymous From Header

Set to yes to show the caller ID from the SIP INVITE message “From” header, even if the call is an anonymous call. When the parameter is set to no, the phone displays "Anonymous Caller" as the caller ID.

Disable Local Name To Header

The options are No and Yes:

• If No is selected, no changes are made. The default value is No.
• If Yes is selected, the following happens:

– Disables the display name in “Directory” and “Call History” in the “To” header during an outgoing call.

– Ignores the CLID in the “UPDATE” message.

– Redial list is populated based on 18x or 200 OK PAID header with or without Display Name.

Note This field is supported in Firmware Release 7.6.2 and later.

SIP T1

RFC 3261 T1 value (RTT estimate) that can range from 0 to 64 seconds. Defaults to.5 seconds.

SIP T2

RFC 3261 T2 value (maximum retransmit interval for non-INVITE requests and INVITE responses) that can range from 0 to 64 seconds. Defaults to 4 seconds.

SIP T4

RFC 3261 T4 value (maximum duration a message remains in the network) that can range from 0 to 64 seconds. Defaults to 5 seconds.

SIP Timer B

INVITE time-out value that can range from 0 to 64 seconds. Defaults to 16 seconds.

SIP Timer F

Non-INVITE time-out value that can range from 0 to 64 seconds. Defaults to 16 seconds.

SIP Timer H

INVITE final response, time-out value that can range from 0 to 64 seconds. Defaults to 16 seconds.

SIP Timer D

ACK hang-around time that can range from 0 to 64 seconds. Defaults to 16 seconds.

SIP Timer J

Non-INVITE response hang-around time that can range from 0 to 64 seconds. Defaults to 16 seconds.

INVITE Expires

INVITE request Expires header value. If you enter 0, the Expires header is not included in the request. Ranges from 0 to 19999999999999999999999999999999.

Defaults to 240 seconds.

ReINVITE Expires

ReINVITE request Expires header value. If you enter 0, the Expires header is not included in the request. Ranges from 0 to 19999999999999999999999999999999.

Defaults to 30

Reg Min Expires

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

Defaults to 1 second.

Reg Max Expires

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

Defaults to 7200.

Reg Retry Intvl (See note below)

Interval to wait before the SPA9000 retries registration after failing during the last registration.

Defaults to 30.

Reg Retry Long Intvl (See note below)

When registration fails with a SIP response code that does not match<Retry Reg RSC>, the SPA9000 waits for the specified length of time before retrying. If this interval is 0, the SPA9000 stops trying. This value should be much larger than the Reg Retry Intvl value, which should not be 0.

Defaults to 1200.

Reg Retry Random Delay

Random delay range (in seconds) to add to <Register Retry Intvl> when retrying REGISTER after a failure. This feature was added in Release 5.1.

Defaults to blank, which disables this feature.

Reg Retry Long Random Delay

Random delay range (in seconds) to add to <Register Retry Long Intvl> when retrying REGISTER after a failure. This feature was added in Release 5.1.

Defaults to blank, which disables this feature.

Reg Retry Intvl Cap

The maximum value to cap the exponential back-off retry delay (which starts at <Register Retry Intvl> and doubles on every REGISTER retry after a failure). In other words, the retry interval is always at <Register Retry Intvl> seconds after a failure. If this feature is enabled, <Reg Retry Random Delay> is added on top of the exponential back-off adjusted delay value. This feature was added in Release 5.1.

Defaults to blank, which disables the exponential back-off feature.

Sub Min Expires

This value sets the lower limit of the REGISTER expires value returned from the Proxy server.

Sub Max Expires

This value sets the upper limit of the REGISTER min-expires value returned from the Proxy server in the Min-Expires header. Defaults to 7200.

Sub Retry Intvl

This value (in seconds) determines the retry interval when the last Subscribe request fails. Defaults to 10.

SIT1 RSC

SIP response status code for the appropriate Special Information Tone (SIT). For example, if you set the SIT1 RSC to 404, when the user makes a call and a failure code of 404 is returned, the SIT1 tone is played. Reorder or Busy Tone is played by default for all unsuccessful response status code for SIT 1 RSC through SIT 4 RSC.
Defaults to blank.

SIT2 RSC

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

SIT3 RSC

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

SIT4 RSC

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

Try Backup RSC

This parameter may be set to invoke failover upon receiving specified response codes.

Defaults to blank.

Retry Reg RSC

Interval to wait before the SPA9000 retries registration after failing during the last registration.
Defaults to blank.

RTP Port Min

Minimum port number for RTP transmission and reception. Minimum port number for RTP transmission and reception. Should define a range that contains at least 10 even number ports (twice the number of lines); for example, configure RTP port min to 16384 and RTP port max to 16402.

Defaults to 16384.

RTP Port Max

Maximum port number for RTP transmission and reception. Should define a range that contains at least 10 even number ports (twice the number of lines); for example, configure RTP port min to 16384 and RTP port max to 16402.

Defaults to 16482.

RTP Packet Size

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

Defaults to 0.030.

Max RTP ICMP Err

Number of successive ICMP errors allowed when transmitting RTP packets to the peer before the SPA9000 terminates the call. If value is set to 0, the SPA9000 ignores the limit on ICMP errors.

Defaults to 0.

RTCP Tx Interval

Interval for sending out RTCP sender reports on an active connection. It can range from 0 to 255 seconds. During an active connection, the SPA9000 can be programmed to send out compound RTCP packet on the connection. Each compound RTP packet except the last one contains a SR (Sender Report) and a SDES (Source Description). The last RTCP packet contains an additional BYE packet. Each SR except the last one contains exactly 1 RR (Receiver Report); the last SR carries no RR. The SDES contains CNAME, NAME, and TOOL identifiers. The CNAME is set to <User ID>@<Proxy>, NAME is set to <Display Name> (or Anonymous if user blocks caller ID), and TOOL is set to the Vendor/Hardware-platform-software-version (such as Cisco/SPA9000-1.0.31(b)). The NTP timestamp used in the SR is a snapshot of the SPA9000 local time, not the time reported by an NTP server. If the SPA9000 receives a RR from the peer, it attempts to compute the round trip delay and show it as the <Call Round Trip Delay> value (ms) in the Info section of SPA9000 web page.

Defaults to 0.

No UDP Checksum

Select yes if you want the SPA9000 to calculate the UDP header checksum for SIP messages. Otherwise, select no.

Defaults to no.

Symmetric RTP

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

Defaults to no.

Stats In BYE

Determines whether the IP phone includes the P-RTP-Stat header or response to a BYE message. The header contains the RTP statistics of the current call. Select yes or no from the drop-down menu. The format of the P-RTP-Stat header is:

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

Defaults to no.

AVT Dynamic Payload

AVT dynamic payload type. Ranges from 96-127.

Defaults to 101.

INFOREQ Dynamic Payload

INFOREQ dynamic payload type.

Defaults to blank.

G726r16 Dynamic Payload

G.726-16 dynamic payload type. Ranges from 96-127.

Defaults to 98.

Note Not applicable to Cisco SPA525G or Cisco SPA525G2 or Cisco WIP310.

G726r24 Dynamic Payload

G.726-24 dynamic payload type. Ranges from 96-127.

Defaults to 97.

Note Not applicable to Cisco SPA525G or Cisco SPA525G2 or Cisco WIP310.

G726r32 Dynamic Payload

G726r32 dynamic payload type.

The default is 2.

G726r40 Dynamic Payload

G.726-40 dynamic payload type. Ranges from 96-127.

Defaults to 96.

Note Not applicable to Cisco SPA525G or Cisco SPA525G2 or Cisco WIP310.

G729b Dynamic Payload

G729b Dynamic Payload type.

Defaults to 99.

EncapRTP Dynamic Payload

EncapRTP Dynamic Payload type.

Defaults to 112.

RTP-Start-Loopback Dynamic

RTP-Start-Loopback Dynamic Payload.

Defaults to 113.

RTP-Start-Loopback Codec

RTP-Start-Loopback Codec. Select one of following: G711u, G711a, G726-16, G726-24, G726-32, G726-40, G729a, or G723.

Cisco SPA525G or Cisco SPA525G2: G711u, G711a, G726-32, G729a, G722.

Defaults to G711u.

AVT Codec Name

AVT codec name used in SDP.

Defaults to telephone-event.

G711u Codec Name

G.711u codec name used in SDP.

Defaults to PCMU.

G711a Codec Name

G.711a codec name used in SDP.

Defaults to PCMA.

G726r16 Codec Name

G.726-16 codec name used in SDP.

Defaults to G726-16.

Note Not applicable to Cisco SPA525G or Cisco SPA525G2 or Cisco WIP310.

G726r24 Codec Name

G.726-24 codec name used in SDP.

Defaults to G726-24.

Note Not applicable to Cisco WIP310, Cisco SPA525G or Cisco SPA525G2.

G726r32 Codec Name

G.726-32 codec name used in SDP.

Defaults to G726-32.

G726r40 Codec Name

G.726-40 codec name used in SDP.

Defaults to G726-40.

Note Not applicable to Cisco SPA525G or Cisco SPA525G2 or Cisco WIP310.

G729a Codec Name

G.729a codec name used in SDP.

Defaults to G729a.

G729b Codec Name

G.729b codec name used in SDP.

Defaults to G729ab.

G722 Codec Name

G.722 codec name used in SDP.

Defaults to G722.

Note Not supported on the Cisco WIP310.

EncapRTP Codec Name

EncapRTP codec name used in SDP.

Defaults to encaprtp.

Handle VIA received

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

Defaults to no.

Handle VIA rport

If you select yes, the SPA9000 processes the rport parameter in the VIA header (this is inserted by the server in a response to any of its requests). If you select no, the parameter is ignored. Select yes or no from the drop-down menu.

Defaults to no.

Insert VIA received

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

Defaults to no.

Insert VIA rport

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

Defaults to no.

Substitute VIA Addr

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

Defaults to no.

Send Resp To Src Port

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

Defaults to no.

STUN Enable

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

Defaults to no.

STUN Test Enable

If the STUN Enable feature is enabled and a valid STUN server is available, the SPA9000 can perform a NAT-type discovery operation when it powers on. It contacts the configured STUN server, and the result of the discovery is reported in a Warning header in all subsequent REGISTER requests. If the SPA9000 detects symmetric NAT or a symmetric firewall, NAT mapping is disabled.

Defaults to no.

STUN Server

IP address or fully-qualified domain name of the STUN server to contact for NAT mapping discovery. You can use a public STUN server or set up your own STUN server.

EXT IP

External IP address to substitute for the actual IP address of the SPA9000 in all outgoing SIP messages. If 0.0.0.0 is specified, no IP address substitution is performed.

If this parameter is specified, the SPA9000 assumes this IP address when generating SIP messages and SDP (if NAT Mapping is enabled for that line). However, the results of STUN and VIA received parameter processing, if available, supersede this statically configured value.

Defaults to blank.

EXT RTP Port Min

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

Defaults to blank.

NAT Keep Alive Intvl

Interval between NAT-mapping keep alive messages.

Defaults to 15.

Redirect Keep Alive

If enabled, the IP phone redirects the keepalive message when SIP_301_MOVED_PERMANENTLY is received as the registration response.

Linksys Key System

Enable or disable the Linksys Key System on the IP phone.

Defaults to yes.

Multicast Address

The multicast address is used by the Cisco SPA9000 to communicate with the Cisco SPA IP phones.

Defaults to 224.168.168.168:6061.

Key System Auto Discovery

Enables or disables auto discovery of the call control server (for example, the Cisco SPA9000). Disable this feature for teleworkers or other scenarios where multicast does not work.

Key System IP Address

IP address of the call control server IP. Enter the IP address for teleworkers or other scenarios where multicast does not work.

Force LAN Codec

The choices are: none, G.711u, or G.711a. Defaults to none.

Auto Ans GrPage On Active Call

Used with the Cisco UC320. Allows you to enable or disable “auto answer” of a group page when there is an active call on the phone. When set to yes (the default), a group page is automatically answered by the phone even if the user is on an active call. When set to no, the page is not automatically answered, and the user can decide to answer the page or stay on the active call.

Dial Tone

Prompts the user to enter a phone number.

Defaults to 350@-19,440@-19;10(*/0/1+2).

Bluetooth Dial Tone (Cisco SPA525G or Cisco SPA525G2 only)

Indicates a bluetooth headset is paired and the user can make a call.

Defaults to 350@-19,440@-19;1(0/*/0);10(*/0/1+2).

Outside Dial Tone

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

Defaults to 420@-16;10(*/0/1).

Prompt Tone

Prompts the user to enter a call forwarding phone number.

Defaults to 520@-19,620@-19;10(*/0/1+2).

Busy Tone

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

Defaults to 480@-19,620@-19;10(.5/.5/1+2).

Reorder Tone

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

Defaults to 480@-19,620@-19;10(.25/.25/1+2).

Off Hook Warning Tone

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

Defaults to 480@10,620@0;10(.125/.125/1+2).

Ring Back Tone

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

Defaults to 440@-19,480@-19;*(2/4/1+2).

Call Waiting Tone

Played when a call is waiting. Defaults to
440@-10;30(.3/9.7/1)

Confirm Tone

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

Defaults to 600@-16; 1(.25/.25/1).

SIT1 Tone

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

Defaults to 985@-16,1428@-16,1777@-16;20(.380/0/1,.380/0/2,.380/0/3,0/4/0).

SIT2 Tone

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

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

SIT3 Tone

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

Defaults to 914@-16,1371@-16,1777@-16;20(.380/0/1,.380/0/2,.380/0/3,0/4/0)

SIT4 Tone

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

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

MWI Dial Tone

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

Defaults to 350@-19,440@-19;2(.1/.1/1+2);10(*/0/1+2).

Cfwd Dial Tone

Played when all calls are forwarded.

Defaults to 350@-19,440@-19;2(.2/.2/1+2);10(*/0/1+2).

Holding Tone

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

Defaults to 600@-19*(.1/.1/1,.1/.1/1,.1/9.5/1).

Conference Tone

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

Defaults to 350@-19;20(.1/.1/1,.1/9.7/1).

Secure Call Indication Tone

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

Defaults to 397@-19,507@-19;15(0/2/0,.2/.1/1,.1/2.1/2).

Page Tone

Specifies the tone transmitted when the paging feature is enabled.

Defaults to 600@-16;.3(.05/0.05/1).

Alert Tone

Played when an alert occurs.

Defaults to 600@-19;.2(.05/0.05/1).

System Beep

Audible notification tone played when a system error occurs.

Defaults to 600@-16;.1(.05/0.05/1).

Note Applies to Cisco SPA525G or Cisco SPA525G2 or Cisco WIP310 only.

Call Pickup Tone

This field provides the ability to configure an audio indication for call pickup. The call pickup tone is the tone to play when the BLF button is configured and there is an incoming alert on the button. The default value for this parameter is
440@-10;30(.3/9.7/1), which is the same as the call waiting tone.

This feature appears as follows in the phone configuration file:

Cadence 1

Cadence script for distinctive ring 1.

Defaults to 60(2/4).

Cadence 2

Cadence script for distinctive ring 2.

Defaults to 60(.3/.2, 1/.2,.3/4).

Cadence 3

Cadence script for distinctive ring 3.

Defaults to 60(.8/.4,.8/4).

Cadence 4

Cadence script for distinctive ring 4.

Defaults to 60(.4/.2,.3/.2,.8/4).

Cadence 5

Cadence script for distinctive ring 5.

Defaults to 60(.2/.2,.2/.2,.2/.2,1/4)

Cadence 6

Cadence script for distinctive ring 6.

Defaults to 60(.2/.4,.2/.4,.2/4).

Cadence 7

Cadence script for distinctive ring 7.

Defaults to 60(4.5/4).

Cadence 8

Cadence script for distinctive ring 8.

Defaults to 60(0.25/9.75)

Cadence 9

Cadence script for distinctive ring 9.

Defaults to 60(.4/.2,.4/2).

Reorder Delay

Delay after far end hangs up before reorder (busy) tone is played. 0 = plays immediately, inf = never plays. Range: 0–255 seconds. Set to 255 to return the phone immediately to on-hook status and to not play the tone.

Defaults to 5.

Call Back Expires

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

Defaults to 1800.

Call Back Retry Intvl

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

Defaults to 30.

Call Back Delay

Delay after receiving the first SIP 18x response before declaring the remote end is ringing. If a busy response is received during this time, the SPA9000 still considers the call as failed and keeps on retrying.

Defaults to 0.5.

Interdigit Long Timer

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

Defaults to 10.

Interdigit Short Timer

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

Defaults to 3.

Call Return Code

This code calls the last caller.

Defaults to *69.

Blind Transfer Code

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

Defaults to *98.

Call Back Act Code

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

Defaults to *66.

Call Back Deact Code

Cancels a callback.

Defaults to *86.

Cfwd All Act Code

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

Defaults to *72.

Cfwd All Deact Code

Cancels call forwarding of all calls.

Defaults to *73.

Cfwd Busy Act Code

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

Defaults to *90.

Cfwd Busy Deact Code

Cancels call forwarding of busy calls.

Defaults to *91.

Cfwd No Ans Act Code

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

Defaults to *92.

Cfwd No Ans Deact Code

Cancels call forwarding of no-answer calls.

Defaults to *93.

CW Act Code

Enables call waiting on all calls.

Defaults to *56.

CW Deact Code

Disables call waiting on all calls.

Defaults to *57.

CW Per Call Act Code

Enables call waiting for the next call.

Defaults to *71.

CW Per Call Deact Code

Disables call waiting for the next call.

Defaults to *70.

Block CID Act Code

Blocks caller ID on all outbound calls.

Defaults to *67.

Block CID Deact Code

Removes caller ID blocking on all outbound calls.

Defaults to *68.

Block CID Per Call Act Code

Blocks caller ID on the next outbound call.

Defaults to *81.

Block CID Per Call Deact Code

Removes caller ID blocking on the next inbound call.

Defaults to *82.

Block ANC Act Code

Blocks all anonymous calls.

Defaults to *77.

Block ANC Deact Code

Removes blocking of all anonymous calls.

Defaults to *87.

DND Act Code

Enables the do not disturb feature.

Defaults to *78.

DND Deact Code

Disables the do not disturb feature.

Defaults to *79.

Secure All Call Act Code

Makes all outbound calls secure.

Defaults to *16.

Secure No Call Act Code

Makes all outbound calls not secure.

Defaults to *17.

Secure One Call Act Code

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

Defaults to *18.

Secure One Call Deact Code

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

Defaults to *19.

Paging Code

The star code used for paging the other clients in the group.

Defaults to *96.

Call Park Code

The star code used for parking the current call.

Defaults to *38.

Call Pickup Code

The star code used for picking up a ringing call.

Defaults to *36.

Call UnPark Code

The star code used for picking up a call from the call park.

Defaults to *39.

Group Call Pickup Code

The star code used for picking up a group call.

Defaults to *37.

Media Loopback Code

The star code used for media loopback.

Defaults to *03.

Referral Services Codes

These codes tell the SPA9000 what to do when the user places the current call on hold and is listening to the second dial tone.

One or more *code can be configured into this parameter, such as *98, or *97|*98|*123, etc. Max total length is 79 chars. This parameter applies when the user places the current call on hold (by Hook Flash) and is listening to second dial tone. Each *code (and the following valid target number according to current dial plan) entered on the second dial-tone triggers the SPA9000 to perform a blind transfer to a target number that is prepended by the service *code.

For example, after the user dials *98, the SPA9000 plays a special dial tone called the Prompt Tone while waiting for the user the enter a target number (which is checked according to dial plan as in normal dialing). When a complete number is entered, the SPA9000 sends a blind REFER to the holding party with the Refer-To target equals to *98<target_number>. This feature allows the Cisco SPA9000 to hand off a call to an application server to perform further processing, such as call park.

The *codes should not conflict with any of the other vertical service codes internally processed by the Cisco SPA9000. You can empty the corresponding *code that you do not want to Cisco SPA9000 to process.

Feature Dial Services Codes

These codes tell the Cisco SPA9000 what to do when the user is listening to the first or second dial tone.

One or more *code can be configured into this parameter, such as *72, or *72|*74|*67|*82, and so forth. The maximum total length is 79 characters. This parameter applies when the user has a dial tone (first or second dial tone). Enter *code (and the following target number according to current dial plan) entered at the dial tone triggers the Cisco SPA9000 to call the target number prepended by the *code. For example, after user dials *72, the Cisco SPA9000 plays a prompt tone awaiting the user to enter a valid target number. When a complete number is entered, the Cisco SPA9000 sends a INVITE to *72<target_number> as in a normal call. This feature allows the proxy to process features like call forward (*72) or BLock Caller ID (*67).

Feature Dial Services Codes (continued

The *codes should not conflict with any of the other vertical service codes internally processed by the Cisco SPA9000. You can empty the corresponding *code that you do not want to Cisco SPA9000 to process.

You can add a parameter to each *code in Features Dial Services Codes to indicate what tone to play after the *code is entered, such as *72‘c‘|*67‘p‘. Below are a list of allowed tone parameters (note the use of back quotes surrounding the parameter without spaces)

• c = Cfwd Dial Tone
• d = Dial Tone
• m = MWI Dial Tone
• o = Outside Dial Tone
• p = Prompt Dial Tone
• s = Second Dial Tone
• x = No tones are place, x is any digit not used above

If no tone parameter is specified, the Cisco SPA9000 plays Prompt tone by default.

If the *code is not to be followed by a phone number, such as *73 to cancel call forwarding, do not include it in this parameter. In that case, simple add that *code in the dial plan and the SPA9000 send INVITE *73@..... as usual when user dials *73.

Prefer G711u Code

Makes this codec the preferred codec for the associated call.

Defaults to *017110.

Force G711u Code

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

Defaults to *027110.

Prefer G711a Code

Makes this codec the preferred codec for the associated call.

Defaults to *017111

Force G711a Code

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

Defaults to *027111.

Prefer G722 Code

Makes this codec the preferred codec for the associated call.

Defaults to *01722.

Only one G.722 call at a time is allowed. If a conference call is placed, a SIP re-invite message is sent to switch the calls to narrowband audio.

Note Not supported on the Cisco WIP310.

Force G722 Code

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

Defaults to *02722.

Only one G.722 call at a time is allowed. If a conference call is placed, a SIP re-invite message is sent to switch the calls to narrowband audio.

Note Not supported on the Cisco WIP310.

Prefer L16 Code

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

Defaults to *01016.

Force L16 Code

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

Defaults to *02016.

Prefer G723 Code

Makes this codec the preferred codec for the associated call.

Defaults to *01723.

Note Not applicable to Cisco WIP310, Cisco SPA525G or Cisco SPA525G2.

Force G723 Code

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

Defaults to *02723.

Note Not applicable to Cisco WIP310, Cisco SPA525G or Cisco SPA525G2.

Prefer G726r16 Code

Makes this codec the preferred codec for the associated call.

Defaults to *0172616.

Note Not applicable to Cisco WIP310, Cisco SPA525G or Cisco SPA525G2.

Force G726r16 Code

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

Defaults to *0272616.

Note Not applicable to Cisco WIP310, Cisco SPA525G or Cisco SPA525G2.

Prefer G726r24 Code

Makes this codec the preferred codec for the associated call.

Defaults to *0172624.

Note Not applicable to Cisco WIP310, Cisco SPA525G or Cisco SPA525G2.

Force G726r24 Code

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

Defaults to *0272624.

Note Not applicable to Cisco WIP310, Cisco SPA525G or Cisco SPA525G2.

Prefer G726r32 Code

Makes this codec the preferred codec for the associated call.

Defaults to *0172632.

Force G726r32 Code

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

Defaults to *0272632.

Prefer G726r40 Code

Makes this codec the preferred codec for the associated call.

Defaults to *0172640.

Note Not applicable to Cisco WIP310, Cisco SPA525G or Cisco SPA525G2.

Force G726r40 Code

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

Defaults to *0272640.

Note Not applicable to Cisco WIP310, Cisco SPA525G or Cisco SPA525G2.

Prefer G729a Code

Makes this codec the preferred codec for the associated call.

Defaults to *01729.

Force G729a Code

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

Defaults to *02729.

Time Zone

Selects the number of hours to add to GMT to generate the local time for caller ID generation. Choices are GMT-12:00, GMT-11:00,…, GMT, GMT+01:00, GMT+02:00, …, GMT+13:00.

Defaults to GMT-08:00.

Time Offset

This specifies the offset from GMT to use for the local system time.

Ignore DHCP Time Offset

See Ignore DHCP Time Offset in Miscellaneous.

Daylight Saving Time Rule

See Daylight Saving Time Rule in Miscellaneous.

Daylight Saving Time Enable

Select yes to enable Daylight Saving Time.

Dictionary Server Script.

See Dictionary Server Script in Miscellaneous.

Language Selection

See Language Selection in Miscellaneous.

Set Local Date (mm/dd)

Sets the local date (mm represents the month and dd represents the day). The year is optional and uses two or four digits.

Note Not applicable to the Cisco SPA525G or Cisco SPA525G2.

Set Local Time (HH/mm)

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

Note Not applicable to the Cisco SPA525G or Cisco SPA525G2.

Time Zone

Selects the number of hours to add to GMT to generate the local time for caller ID generation. Choices are GMT-12:00, GMT-11:00,…, GMT, GMT+01:00, GMT+02:00, …, GMT+13:00.

Defaults to GMT-08:00.

Note Found in the Time section for the Cisco SPA525G or Cisco SPA525G2.

Time Offset (HH/mm)

This specifies the offset from GMT to use for the local system time.

Note Found in the Time section for the Cisco SPA525G or Cisco SPA525G2.

Ignore DHCP Time Offset

When used with some routers that have DHCP with time offset values configured, the IP phone uses the router settings and ignores the IP phone time zone and offset settings. To ignore the router DHCP time offset value, and use the local time zone and offset settings, choose yes for this option. Choosing no causes the IP phone to use the router’s DHCP time offset value. The default value is yes.

Note Found in the Time section for the Cisco SPA525G or Cisco SPA525G2.

Daylight Saving Time Rule

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

This is the format of the rule: Start = <start-time>; end=<end-time>; save = <save-time>.

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

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

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

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

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

Daylight Saving Time Rule (continued)

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

HH stands for hours (0-23).

mm stands for minutes (0-59).

ss stands for seconds (0-59).

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

Note Found in the Time section for the Cisco SPA525G or Cisco SPA525G2.

Daylight Saving Time Enable

Select yes to enable Daylight Saving Time.

Note Found in the Time section for the Cisco SPA525G or Cisco SPA525G2.

DTMF Playback Level

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

Defaults to -16.

DTMF Playback Length

Local DTMF playback duration in milliseconds.

Defaults to.1.

Inband DTMF Boost

Controls the amount of amplification applied DTMF signals.

Choices are OdB, 3dB, 6dB, 9dB, 12dB, 15dB, or 18dB.

Defaults to 12dB.

Dictionary Server Script/SCCP Dictionary Server Script (Cisco SPA525G or Cisco SPA525G2 SCCP only)

Defines the location of the dictionary server, the languages available and the associated dictionary. The syntax is as follows:

<Dictionary_Server_Script ua="na">

</Dictionary_Server_Script>

Defaults to blank and the maximum number of characters is 512. The detailed format is as follows:

serv={server ip port and root path};

d0=<language0>;x0=<dictionary0 filename>;

d1=<language1>;x1=<dictionary1 filename>;

d2=<language2>;x2=<dictionary2 filename>;

d3=<language3>;x3=<dictionary3 filename>;

d4=<language4>;x4=<dictionary4 filename>;

d5=<language5>;x5=<dictionary5 filename>;

d6=<language6>;x6=<dictionary6 filename>;

d7=<language3>;x7=<dictionary7 filename>;

d8=<language8>;x8=<dictionary8 filename>;

d9=<language5>;x9=<dictionary9 filename>;

The following is an example value:

<Dictionary_Server_Script ua="na">

serv=tftp://192.168.1.119/;d0=English;x0=enS_v101.xml;d1=Spanish;x1=esS_v101.xml </Dictionary_Server_Script>

Note Not applicable to the Cisco WIP310.

Language Selection/SCCP Language Selection (Cisco SPA525G or Cisco SPA525G2 SCCP Only)

Specifies the default language. The value needs to match one of the languages supported by the dictionary server. The script (dx value) is as follows:

Defaults to blank and the maximum number of characters is 512. The following is an example:

Note Not applicable to the Cisco WIP310.

Default Character Encoding (Cisco SPA303, Cisco SPA500 Series)

The default is ISO-8859-1 for backward compatibility with Cisco SPA900 series phones. If set to UTF-8, line keys and other labels entered by using the phone web user interface containing UTF-8 characters are displayed correctly on the phone. (SIP only)

Locale

Choose the locale that should be set in the HTTP Accept-Language header.

Station Display Name

Name to identify the IP phone; appears on the IP phone screen on models that have a display. You can use spaces in this field and the name does not have to be unique. If both the Station Display Name and Station Name fields are populated, the Station Display Name field takes precedence and is displayed on the phone.

Station Name

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

Voice Mail Number

Phone number or URL to check voice mail. The service provider often hosts a voice mail service. The advantages of hosted voice mail include:

• Advanced features such as voice mail to email conversion.
• Calls can go to voice mail when the broadband connection is down.

Text Logo

Text logo to display when the phone boots up. A service provider, for example, can enter logo text as follows:

• Up to 2 lines of text
• Each line must be fewer than 32 characters
• Insert a new line character (\n) between lines
• Insert escape code %0a

For example, Super\n%0aTelecom displays:

Use the + character to add spaces for formatting. For example, you can add multiple + characters before and after the text to center it.

Note Not applicable to the Cisco WIP310, Cisco SPA301, or Cisco SPA501G. On the Cisco SPA525G or Cisco SPA525G2, this setting is located on the User tab.

BMP Picture Download URL

URL locating the bitmap (.BMP) file to display on the IP phone screen background.

For more information, see Configuring Phone Information and Display Settings .

Note Not applicable to the Cisco WIP310, Cisco SPA301, or Cisco SPA501G. On the Cisco SPA525G or Cisco SPA525G2, this setting is located on the User tab.

Select Logo

Select from Default, BMP Picture, Text Logo, or None.

Defaults to Default.

Note Not applicable to the Cisco WIP310, Cisco SPA301, or Cisco SPA501G. On the Cisco SPA525G or Cisco SPA525G2, this setting is located on the User tab.

Select Background Picture

Select from Default, BMP Picture, or None.

Defaults to Default.

Note Not applicable to the Cisco WIP310, Cisco SPA301, or Cisco SPA501G. On the Cisco SPA525G or Cisco SPA525G2, this setting is located on the User tab.

Softkey Labels Font

Choose the font width for the softkey labels to display on your phone. See Customizing Phone Softkeys.

Note Not applicable to the Cisco WIP310, Cisco SPA301, or Cisco SPA501G.

Screen Saver Enable

Enables a screen saver on the IP phone screen. When the phone is idle for a specified time, it enters screen saver mode. (Users can set up screen savers directly using phone Setup button.) When the screen saver is active, the screen shows “Press any key to unlock your phone.”

User input is accepted if the screen saver is active (for example, if a user begins to dial a number while the screen saver is active, the key press is accepted). If the phone LCD is displaying the Home screen and the screen saver becomes active, and if there are no active calls, the following events generated by user input are passed to the Home screen:

• Numeric keys
• Line keys
• Speaker key
• Headset key
• Mail Box key
• Handset off hook

All other key events are not passed.

Note Not applicable to the Cisco WIP310, Cisco SPA301, or Cisco SPA501G. On the Cisco SPA525G or Cisco SPA525G2, this setting is located on the User tab.

Screen Saver Wait

Amount of idle time before screen saver displays.

Note Not applicable to the Cisco WIP310, Cisco SPA301, or Cisco SPA501G. On the Cisco SPA525G or Cisco SPA525G2, this setting is located on the User tab.

Screen Saver Icon

In screen saver mode, the IP phone screen can display:

• A background picture.
• The station time in the middle of the screen.
• A moving padlock icon. When the phone is locked, the status line displays a scrolling message “Press any key to unlock your phone.”
• A moving phone icon.
• The station date and time in the middle of the screen.

Note Not applicable to the Cisco WIP310, Cisco SPA301, or Cisco SPA501G. On the Cisco SPA525G or Cisco SPA525G2, this setting is located on the User tab.

JPEG Logo Download URL (Cisco SPA525G or Cisco SPA525G2)

URL from which to download a.jpg file for the phone logo display.

JPEG Wallpaper Download URL (Cisco SPA525G or Cisco SPA525G2)

URL from which to download a.jpg file for the phone wallpaper.

Enable SMS

Enables sending and receiving of SMS text messages on the phone.

Note Cisco WIP310 only.

Show DTMF Digits when connected

The phone can be configured to show the digits entered by a local user while a call is connected or is proceeding.

The display options are "No"/"Yes"/"Masked".

'No'— No digits are displayed.

'Yes'— Upto 15 most recently dialed digits are displayed.

‘Masked—Behaves like 'Yes,' but the digits are masked with '* 's.

Note Display location of DTMF digits is slightly different on Cisco SPA5xx and Cisco SPA525 phones.

SPA5xx : Dialed digits are displayed on the second line of the call screen. If the call is diverted while a new digit is entered, the line switches from "Via: XXXX" to digits for 10 seconds.

If no digits are entered during the last 10 seconds, the line will switch back to "Via:XXXX."

SPA525 : Dialed digits are displayed on the fourth line. For a conference call, two call appearances present on the GUI, and the digits are displayed only on the highlighted call.

Extension

Extension number of the line key.

Short Name

A short label shown on the IP phone screen for Line Key 1 through Line Key 6.

Share Call Appearance

Yes indicates that Line Key 1/2/3/4/5/6 is a shared call appearance. Otherwise this call appearance is not shared (it is private).

Defaults to no.

Extended Function

Use to assign Busy Lamp Field, Call Pickup, and Speed Dial Functions to Idle Lines on the IP phone.

Syntax is:

where:

• fnc: function
• blf: busy lamp field
• cp: call pickup
• sub: station name (not needed for speed dial)
• ext or usr: extension or user (the usr and ext keywords are interchangeable)

Note Not applicable to the Cisco WIP310, Cisco SPA301, or Cisco SPA501G.

Line ID Mapping

Specifies the call appearance line ID mapping when the call appearance per line is set to two. Applied to both shared line and private lines. Effective for incoming calls only.

Choose Vertical First or Horizontal First. Each LED can hold multiple calls and the first call on an LED makes it light up. Horizontal first means the second incoming call makes the same line LED flash. Vertical first means the second incoming call lights up the next line LED with the same account.

For example, if Extension 101 is assigned to two LEDs, and Vertical First is selected, the second incoming call on Extension 101 lights up the second LED. The third call makes the first LED flash, and the fourth call makes the second LED flash.

If Horizontal First is selected, the second incoming call on Extension 101 makes the first LED flash. The third call lights up the second LED, and the fourth call makes the second LED flash.

SCA Barge-In Enable

Enables the SCA Barge-In.

Defaults to no.

SCA Sticky Auto Line Seize

When enabled, taking the phone off-hook will not automatically pick up an incoming call on a shared line.

SCA Unseize Delay

Allows shared line unseize delay to be Configurable. Default is 0.

Line Navigation

(Does not apply to the Cisco SPA525G or Cisco SPA525G2.)

This parameter controls the way that the IP phone navigates between call appearances when the user presses the up or down navigation button. The default value is “per line.”

When Per Call is configured, and the user presses the up and down navigation button, the phone displays each call for a line before moving to subsequent lines. For example, if there are two calls on Line 1, pressing the up or down navigation button scrolls between the two calls first, before moving to display any calls that are on Line 2. We recommend you enable programmable softkeys when choosing this option.

When Per Line is configured, and the user presses the up and down navigation button, the phone displays one call appearance for a line, before moving to the next line. For example, if there are two calls on Line 1, pressing the up or down navigation button moves directly to display any calls on Line 2 (rather than the second call on Line 1).

Call Appearance Per Line

This parameter allows you to choose the number of calls per line button. You can choose a value from 2 (the default) to 10.

When you increase the number of calls per line to a value greater than 2, the phone automatically sets the Line ID Mapping option to Horizontal First. Changing the Line ID Mapping value has no effect if the value of the Call Appearance Per Line setting is greater than 2.

Note The maximum number of calls per an SPA525G or 525G2 phone is 10. When the maximum numbers of calls per phone is reached, the phone does not allow you to make a new call and rejects incoming calls.

This parameter is not supported on the Cisco SPA501G and Cisco SPA301 phones. Also, this parameter is only supported when the phones are operating in the SIP mode. In version 7.4.8a, only private call appearances are supported. In versions 7.5.1a and higher private and shared call appearances are supported.

Softkey Navigation Style

If scrollable is chosen, users must use the navigation button on the IP phone to scroll to see the additional softkeys. If more is chosen, a “More” softkey is displayed, which, when pressed, displays the additional softkeys.

Note The Cisco SPA525G and Cisco SPA525G2 always display the “More” softkey when in an idle state.

Idle LED

The call appearance is not is in use and is available to make a new call. Leaving this entry blank indicates the default value of c=g.

Remote Undefined LED

The shared call state is undefined (the phone is still waiting for the state information from the application server). Not applicable if the call appearance is not shared. Leaving this entry blank indicates the default value of c=r;p=d.

Local Seized LED

This phone has seized the call appearance to prepare for a new outbound call. Leaving this entry blank indicates the default value of c=r.

Remote Seized LED

The shared call appearance is seized by another phone. Not application if the call appearance is not shared. Leaving this entry blank indicates the default value of c=r;p=d.

Local Progressing LED

This phone is attempting on this call appearance an outgoing call that is in proceeding (i.e. the called number is ringing). Leaving this entry blank indicates the default value of c=r.

Remote Progressing LED

Another phone is attempting on this shared call appearance an outbound call that is progressing. Not applicable if the call appearance is not shared. Leaving this entry blank indicates the default value of c=r;p=d.

Local Ringing LED

The call appearance is ringing. Leaving this entry blank indicates the default value of c=r;p=f.

Remote Ringing LED

The shared call appearance is in ringing on another phone. Not applicable if the call appearance is not shared. Leaving this entry blank indicates the default value of c=r;p=d.

Local Active LED

The call appearance is engaged in an active call. Leaving this entry blank indicates the default value of c=r.

Remote Active LED

Another station is engaged in an active call on this shared call appearance. Not applicable is this call appearance is not shared. Leaving this entry blank indicates the default value of c=r;p=d.

Local Held LED

The call appearance is held by this phone. Leaving this entry blank indicates the default value of c=r;p=s.

Remote Held LED

Another phone has placed this call appearance on hold. Not applicable if the call appearance is not shared. Leaving this entry blank indicates the default value of c=r,p=s.

Register Failed LED

The corresponding extension has failed to register with the proxy server. Leaving this entry blank indicates the default value of c=a.

Disabled LED

The Call Appearance is disabled (not available for any incoming or outgoing call). Leaving this entry blank indicates the default value of c=o.

Registering LED

The corresponding extension is trying to register with the proxy server. Leaving this entry blank indicates the default value of c=r;p=s.

Call Back Active LED

Call Back operation is currently active on and this call appearance is not shared. Leaving this entry blank indicates the default value of c=r;p=s.

Trunk In-Use LED

A shared trunk is in use.

Trunk No Service LED

A shared trunk is not in service.

Trunk Reserved LED

A shared trunk has been reserved.

Conference Serv

Enable/disable Three way conference service.

Defaults to yes.

Attn Transfer Serv

Enable/disable attended-call-transfer service.

Defaults to yes.

Blind Transfer Serv

Enable/disable blind-call-transfer service.

Defaults to yes.

DND Serv

Enable/disable do-not-disturb service.

Defaults to yes.

Block ANC Serv

Enable/disable block-anonymous-call service.

Defaults to yes.

Call Back Serv

Enable/disable call-back (aka. repeating dialing) service.

Defaults to yes.

Block CID Serv

Enable/disable blocking outbound Caller-ID service.

Defaults to yes.

Secure Call Serv

Enable/disable secure-call service.

Defaults to yes.

Cfwd All Serv

Enable/disable call-forward-all service.

Defaults to yes.

Cfwd Busy Serv

Enable/disable call-forward-on-busy service.

Defaults to yes.

Cfwd On No Ans Serv

Enable/disable call-forward-on-no-answer service.

Defaults to yes.

Paging Serv

Enable/disable the paging service.

Defaults to yes.

Call Park Serv

Enable/disable the call park service.

Defaults to yes.

Call Pick Up Serv

Enable/disable the call pickup service.

Defaults to yes.

ACD Login Serv

Enable/disable the ACD Login Service, used for call centers. Typically enabled with the <SIP-B> parameter.

Defaults to no.

Group Call Pick Up Serv

Enable/disable the group call pickup service.

Defaults to yes.

Group Call Pick Up Serv

Enable/disable the group call pickup service.

Defaults to yes.

ACD Ext

The extension used for handling ACD calls. Select from 1, 2, 3, 4, 5, or 6.

Defaults to 1.

Service Annc Serv

Enable/disable sending announcement requests to a customer-supplied announcement server.

Defaults to no.

Web Serv (Cisco SPA525G or Cisco SPA525G2 only)

Enable/disable the web server.

Defaults to yes.

SMS Serv (Cisco SPA525G or Cisco SPA525G2 only)

Enable/disable the SMS text messaging server.

Ring1

Ring tone script for ring 1. Defaults to n=Classic-1;w=3;c=1.

Ring2

Ring tone script for ring 2. Defaults to n=Classic-2;w=3;c=2.

Ring3

Ring tone script for ring 3. Defaults to n=Classic-3;w=3;c=3.

Ring4

Ring tone script for ring 4. Defaults to n=Classic-4;w=3;c=4.

Ring5

Ring tone script for ring 5. Defaults to n=Simple-1;w=2;c=1.

Ring6

Ring tone script for ring 6. Defaults to n=Simple-2;w=2;c=2.

Ring7

Ring tone script for ring 7. Defaults to n=Simple-3;w=2;c=3.

Ring8

Ring tone script for ring 8. Defaults to n=Simple-4;w=2;c=4.

Ring9

Ring tone script for ring 9. Defaults to n=Simple-5;w=2;c=5.

Ring10

Ring tone script for ring 10. Defaults to n=Office;w=4;c=1.

Ring11

(Cisco SPA300 Series and Cisco SPA500 Series)
Ring tone script for ring 11. Defaults to n=Pulse;w=5;c=1.

(Cisco SPA525G or Cisco SPA525G2)
Ring tone script for ring 11. Defaults to n=Pulse;w=file://Pulse1.raw;c=1.

Ring12

(Cisco SPA300 Series and Cisco SPA500 Series)
Ring tone script for ring 12. Defaults to n=Du-dut;w=6;c=1.

(Cisco SPA525G or Cisco SPA525G2)
Ring tone script for ring 11. Defaults to n=Du-dut;w=file://Ring7.raw;c=1.

Silent Ring Duration

(Cisco SPA 300 Series and Cisco SPA500 Series)
Controls the duration of the silent ring.

For example, if the parameter is set to "20" seconds, the phone plays the silent ring for 20 seconds then sends 480 response to INVITE message.

Warble

(Cisco SPA300 Series and Cisco SPA500 Series)
w=7

((Cisco SPA525G or Cisco SPA525G2))
w=file://Warble.raw

Low

(Cisco SPA300 Series and Cisco SPA500 Series)
w=8

((Cisco SPA525G or Cisco SPA525G2))
w=file://Low.raw

Floor

(Cisco SPA300 Series and Cisco SPA500 Series)
w=9

((Cisco SPA525G or Cisco SPA525G2))
w=file://Floor.raw

Reverb

(Cisco SPA300 Series and Cisco SPA500 Series)
w=10

((Cisco SPA525G or Cisco SPA525G2))
w=file://Reverb.raw

Keypad Tone

Select yes to enable the keypad tone to be played when a key on the keypad is pressed. Select no to silence the keypad.

Keypad Tone Volume

Corresponds to the volume of the keypad tone. Default is 5.

Handset Input Gain

The amount of amplification to apply to the audio input signal for the handset.

Defaults to zero.

Headset Input Gain

The amount of amplification to apply to the audio input signal for the headset.

Defaults to zero.

Note Not applicable to the Cisco SPA301 or the Cisco SPA501G.

Speakerphone Input Gain

The amount of amplification to apply to the audio input signal for the speakerphone.

Defaults to zero.

Note Not applicable to the Cisco SPA301 or the Cisco SPA501G.

Bluetooth Input Gain (Cisco SPA525G or Cisco SPA525G2 only)

The amount of amplification to apply to the audio input signal for the Bluetooth device.

Defaults to zero.

Handset Additional Input Gain

Applies additional input gain to the handset. The maximum value is 9.

Note Does not apply to the Cisco SPA525G or Cisco SPA525G2.

Headset Additional Input Gain

Applies additional input gain to the headset. The maximum value is 9.

Note Does not apply to the Cisco SPA525G or Cisco SPA525G2, or the Cisco SPA501G.

Speakerphone Additional Input Gain

Applies additional input gain to the speakerphone. The maximum value is 9.

Note Does not apply to the Cisco SPA301 or Cisco SPA525G or Cisco SPA525G2 or Cisco SPA501G.

Bluetooth Additional Input Gain (Cisco SPA525G or Cisco SPA525G2 only)

Applies additional input gain to the Bluetooth device.

Defaults to zero. The maximum value is 9.

Directory Enable

Set to yes to enable BroadSoft directory for the phone user. Defaults to no.

Note Not applicable to the Cisco SPA301 or Cisco SPA501G.

XSI Host Server

Enter the name of the server; for example, xsp.xdp.broadsoft.com.

Note Not applicable to the Cisco SPA301 or Cisco SPA501G.

Directory Name

Name of the directory. Displays on the user phone as a directory choice.

Note Not applicable to the Cisco SPA301 or Cisco SPA501G.

Directory Type

Select the type of BroadSoft directory:

Enterprise (default): Allows users to search on last name, first name, user or group ID, phone number, extension, department, or email address.

Group: Allows users to search on last name, first name, user ID, phone number, extension, department, or email address.

Personal: Allows users to search on last name, first name, or telephone number.

Note Not applicable to the Cisco SPA301 or Cisco SPA501G.

Directory UserID

BroadSoft User ID of the phone user; for example, johndoe@xdp.broadsoft.com.

Note Not applicable to the Cisco SPA301 or Cisco SPA501G.

Directory Password

Alphanumeric password associated with the User ID.

Note Not applicable to the Cisco SPA301 or Cisco SPA501G.

Call Feature Sync Ext

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

This feature is disabled by default.

Choose the extension (1 through 5) that is registered to the BroadSoft server.

Note This parameter was removed in software release 7.4.9, and replaced by the Feature Key Sync parameter in the Extension tab.

LDAP Dir Enable

Choose yes to enable LDAP.

LDAP Corp Dir Name

Enter a free-form text name, such as “Corporate Directory.”

LDAP Server

Enter a fully qualified domain name or IP address of LDAP server, in the following format:

LDAP Auth Method

Select the authentication method that the LDAP server requires. Choices are:

None—No authentication is used between the client and the server.

Simple—The client sends its fully-qualified domain name and password to the LDAP server. Might present security issues.

Digest-MD5—The LDAP server sends authentication options and a token to the client. The client returns an encrypted response that is decrypted and verified by the server.

LDAP Client DN

Enter the distinguished name domain components [dc] ; for example:

If using the default Active Directory schema (Name(cn)->Users->Domain), an example of the client DN follows:

LDAP Username

Enter the username for a credentialed user on the LDAP server.

LDAP Password

Enter the password for the LDAP username.

LDAP Search Base

Specify a starting point in the directory tree from which to search. Separate domain components [dc] with a comma. For example:

LDAP Last Name Filter

This defines the search for surnames [sn], known as last name in some parts of the world. For example, sn:(sn=*$VALUE*). This search allows the provided text to appear anywhere in a name, beginning, middle, or end.

LDAP First Name Filter

This defines the search for the common name [cn]. For example, cn:(cn=*$VALUE*). This search allows the provided text to appear anywhere in a name, beginning, middle, or end.

LDAP Search Item 3

Additional customized search item. Can be blank if not needed.

LDAP Item 3 Filter

Customized filter for the searched item. Can be blank if not needed.

LDAP Search Item 4

Additional customized search item. Can be blank if not needed.

LDAP Item 4 Filter

Customized filter for the searched item. Can be blank if not needed.

LDAP Display Attrs

Format of LDAP results display on phone where:

• a—Attribute name
• cn—Common name
• sn—Surname (last name)
• telephoneNumber—Phone number
• n—Display name

For example, n=Phone causes "Phone:" to be displayed in front of the phone number of an LDAP query result when the detail soft button is pressed.

• t—type

When t=p, that is, t is of type phone number, then the retrieved number can be dialed. Only one number can be made dialable. If two numbers are defined as dialable, only the first number is used. For example, a=ipPhone, t=p; a=mobile, t=p;

This example results in only the IP Phone number being dialable and the mobile number will be ignored.

• p—phone number

When p is assigned to a type attribute, example t=p, then the retrieved number is dialable by the phone.

LDAP Number Mapping

Can be blank if not needed.

Note With the LDAP number mapping you can manipulate the number that was retrieved from the LDAP server. For example, you can append 9 to the number if your dial plan requires a user to enter 9 before dialing. Add the 9 prefix by adding (<:9xx.>) to the LDAP Number Mapping field. For example, 555 1212 would become 9555 1212.

If you do not manipulate the number in this fashion, a user can use the Edit Dial feature to edit the number before dialing out.

XML Directory Service Name

Name of the XML Directory. Displays on the user’s phone as a directory choice.

XML Directory Service URL

URL where the XML Directory is located.

XML Application Service Name

Name of the XML application. Displays on the user’s phone as a web application choice.

XML Application Service URL

URL where the XML application is located.

XML User Name

XML service username for authentication purposes.

XML Password

XML service password for authentication purposes.

Cisco XML EXE Auth Mode

Select the authentication mode for http://ip/CGI/Execute. There are 3 modes:

Trusted: This is the default value.

Local Credential: Use the password to authenticate.

Remote Credential: Use XML User Name and XML Password to authenticate.

Extension Mobility Enable

Enable or disable extension mobility.

Defaults to no (disabled).

EM User Domain

The user domain for extension mobility.

Defaults to blank.

Programmable Softkey Enable

Enables programmable softkeys.

Idle Key List

Softkeys that display when the phone is idle.

Missed Call Key List

Softkeys that display when a call has been missed.

Off Hook Key List

Softkeys that display when the receiver is lifted, or the headphone or speakerphone buttons are pressed.

Dialing Input Key List

Softkeys that display when the user must enter dialing data.

Progressing Key List

Softkeys that display when a call is attempting to connect. (Cisco SPA525G or Cisco SPA525G2 only)

Connected Key List

Softkeys that display when a call is connected.

Start-Xfer Key List

Softkeys that display when a call transfer has been initiated.

Start-Conf Key List

Softkeys that display when a conference call has been initiated.

Conferencing Key List

Softkeys that display when a conference call is in progress.

Releasing Key List

Softkeys that display when a call is disconnecting. (Cisco SPA525G or Cisco SPA525G2 only)

Hold Key List

Softkeys that display when one or more calls are on hold. (Cisco SPA525G or Cisco SPA525G2only)

Ringing Key List

Softkeys that display when a call is incoming.

Shared Active Key List

Softkeys that display when a call is active on a shared line. (Cisco SPA525G or Cisco SPA525G2 only)

Shared Held Key List

Softkeys that display when a call is on hold on a shared line. (Cisco SPA525G or Cisco SPA525G2 only)

PSK1 through PSK16

Programmable softkey fields. Enter a string in these fields to configure softkeys that display on the phone screen. You can create softkeys for speed dials to numbers or extensions, vertical service activation codes (* codes), or XML scripts. For more information on the types of softkeys that can be created and the string syntax to enter, see the “Programmable Softkeys” section.

Record Enable

The Cisco SPA525G2 supports recording a call to a USB flash drive. Select yes to enable this feature. The user can record and play back the call using the phone interface. The following limitations apply:

• Supported in SIP mode only.
• Recording is limited to a single active call. Multiple calls cannot be recorded at the same time.

If the user performs a blind transfer, call recording stops immediately. If the user performs an attended transfer, call recording stops after the local call exits from the bridged call. Call recording stops when a call is parked.

If an incoming call is answered while on the first call, the first call continues to be recorded after the second call is connected. The calls must be joined by conference to record both calls. If a call is put on hold and the user makes another call, the first call continues to be recorded unless the calls are joined by conference. Call recording can only be performed on conferenced calls when the original recorded call is still active, or when the recording is started after both calls are joined by conference.

If call recording is stopped on a call and started on another call, a new, separate call recording file is created.

The remote call number and call name are part of the filename of the record file. If the filename contains any of these characters (\ / : * ? " < >), that character is replaced by '-' in the call recording filename.

Record Beep Reminder

When enabled, the phone plays a beep when call recording is started. This feature is enabled by default.

Share Ext

Indicates whether this extension is to be shared with other Cisco SPA IP phones or private. If the extension is not shared, then a call appearance assigned to this extension is not shared, regardless the setting of <Share Call Appearance> for that call appearance. If the extension is shared, then whether or not a call appearance assigned to this extension is shared follows the setting of <Share Call Appearance> for that call appearance. The choices are shared or private.

Defaults to shared.

Shared User ID

The user identified assigned to the shared line appearance.

Subscription Expires

Number of seconds before the SIP subscription expires. Before the subscription expiration, the phone gets NOTIFY messages from the SIP server on the status of the shared phone extension. Defaults to 60 seconds.

Restrict MWI

When enabled, the message waiting indicator lights only for messages on private lines.

Monitor User ID (Cisco SPA300 Series, Cisco SPA500 Series)

This field is for future use.

NAT Mapping Enable

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

Defaults to no.

NAT Keep Alive Enable

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

Defaults to no.

NAT Keep Alive Msg

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

Defaults to $NOTIFY.

NAT Keep Alive Dest

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

Defaults to $PROXY.

SIP TOS/DiffServ Value

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

Defaults to 0x68.

SIP CoS Value

CoS value for SIP messages.

Defaults to 3.

RTP TOS/DiffServ Value

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

Defaults to 0xb8.

RTP CoS Value

CoS value for RTP data.

Defaults to 6.

Network Jitter Level

Determines how jitter buffer size is adjusted by the Cisco SPA9000. Jitter buffer size is adjusted dynamically. The minimum jitter buffer size is 30 milliseconds or (10 milliseconds + current RTP frame size), whichever is larger, for all jitter level settings. However, the starting jitter buffer size value is larger for higher jitter levels. This setting controls the rate at which the jitter buffer size is adjusted to reach the minimum. Select the appropriate setting: low, medium, high, very high, or extremely high.

Defaults to high.

Jitter Buffer Adjustment

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

Defaults to up and down.

SIP Transport

Select from UDP, TCP, or TLS.

Defaults to UDP.

SIP Port

Port number of the SIP message listening and transmission port.

Defaults to 5060.

SIP 100REL Enable

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

Defaults to no.

EXT SIP Port

The external SIP port number.

Auth Resync-Reboot

If this feature is enabled, the Cisco SPA9000 authenticates the sender when it receives the NOTIFY resync reboot (RFC 2617) message. To use this feature, select yes. Otherwise, select no.

Defaults to yes.

SIP Proxy-Require

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

SIP Remote-Party-ID

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

Defaults to yes.

Referor Bye Delay

Controls when the SPA9000 sends BYE to terminate stale call legs upon completion of call transfers. Multiple delay settings (Referor, Refer Target, Referee, and Refer-To Target) are configured on this screen. For the Referor Bye Delay, enter the appropriate period of time in seconds.

Defaults to 4.

Refer-To Target Contact

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

Defaults to no.

Referee Bye Delay

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

Defaults to 0.

SIP Debug Option

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

none—No logging.

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

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

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

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

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

full—Logs all SIP messages in full text.

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

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

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

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

Defaults to none.

Refer Target Bye Delay

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

Defaults to 0.

Sticky 183

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

Defaults to no.

Auth INVITE

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

Ntfy Refer On 1xx-To-Inv

If set to yes , as a transferee, the phone will send a NOTIFY with Event:Refer to the transferor for any 1xx response returned by the transfer target, on the transfer call leg.

If set to no , the phone will only send a NOTIFY for final responses (200 and higher).

Note Not applicable to the Cisco WIP310.

Use Anonymous With RPID

This parameter applies only if <SIP Remote-Party-ID> is set to yes; otherwise, it is ignored.

If the parameter is set to yes, the FROM header's display-name and user-id fields are set to anonymous when the caller blocks his caller-id. If the parameter is set to no, the FROM header's display-name and user-id are not masked. The Remote-Party-ID header indicates privacy=full when the caller wishes to block his caller-id.

Defaults to yes.

Note Not applicable to the Cisco WIP310.

Set G729annexb

Configure G.729 Annex B settings.

Note Not applicable to Cisco SPA525G or Cisco SPA525G2.

Voice Quality Report Address

For configuration of a SIP event package, SIP PUBLISH, that enables the collection and reporting of metrics that measure the quality for VoIP sessions. Voice call quality information derived from RTCP-XR and call information from SIP is conveyed from a User Agent in a session to the third party in SIP PUBLISH method.

To configure, first configure RTCP-XR (see RTP Parameters). Configure the RTCP Tx Interval. In the Voice Quality Report Address field, enter the name of the collector that collects the statistics from the SIP PUBLISH events. For example, enter
collector@fully-qualified-domain-name (collector@reports.cisco.com) or
collector@IP-address (collector@192.168.5.1).

After RTCP-XR feature is enabled, the call status information is updated on Voice > Info during an active call. Additionally, RTCP-XR packets containing a voice metrics block report are sent with the interval specified in the RTCP Tx Interval. When the call session is ended, a SIP PUBLISH with voice metrics information is sent to the collector endpoint. This parameter supports a full SIP URI. Examples of valid addresses are:

• collector@domain.com
• 123.collect@123.123.123.123:5555
• 5678@domain.com:5656

For example to configure for extension 1, edit the phone configuration file as follows:

User Equal Phone

When a tel URL is converted to a SIP URL and the telephone number is represented by the user portion of the URI, the SIP URL includes the optional :user=phone parameter (RFC3261). For example:

To enable this optional parameter, choose yes. The default value is no.

Blind Attn-Xfer Enable

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

Defaults to no.

MOH Server

User ID or URL of the auto-answering streaming audio server. When only a user ID is specified, the current or outbound proxy is contacted. Music-on-hold is disabled if the MOH Server is not specified.

Defaults to imusic when used with a Cisco SPA9000 IP PBX.

Message Waiting

Indicates whether the Message Waiting Indicator on the phone is lit. This parameter is toggled by a message from the SIP proxy to indicate if a message is waiting. You can manually modify it to clear or set the flag in the Ext 1-6 tab.

Setting this value to Yes can activate stutter tone and VMWI signal. This parameter is stored in long-term memory and survives after reboot or power cycle.

Defaults to No.

Auth Page

Specifies whether to authenticate the invite before auto answering a page.

Defaults to No.

Default Ring

Type of ring heard. This corresponds to the Ring Tone on the Phone tab. Choose from No Ring, 1 through 10, User 1, or User 2.

Defaults to 1.

Auth Page Realm

Identifies the Realm part of the Auth that is accepted when the Auth Page parameter is set to yes. This parameter accepts alphanumeric characters.

Defaults to blank.

Conference Bridge URL

This is the URL used to join into a conference call, generally in the form of the word conference or user@IPaddress:port.

Defaults to blank.

Auth Page Password

Identifies the password used when the Auth Page parameter is set to yes. This parameter accepts alphanumeric characters.

Defaults to blank.

Mailbox ID

Identifies the voice mailbox number/ID for the phone.

Defaults to blank.

Voice Mail Server

Identifies the SpecVM server for the phone, generally the IP address and port number of the VM server.

Voice Mail Subscribe Interval

The expiration time, in seconds, of a subscription to a voice mail server.

State Agent

Reserved feature.

CFWD Notify Serv

Specifies whether to enable a SIP-B feature regarding the sending of a Notify to the phone when a call is forwarded elsewhere.

Defaults to No.

CFWD Notifier

Typically, this field is configured with the SIP proxy information.

User ID with Domain
(Cisco SPA300 Series and SPA 50X IP Phones)

When this field is set to yes, the IP phone will show the Caller ID followed by domain name, and the domain name is also shown in the received calls list. This parameter is used when calls are made between different branches of the same phone system. For example, if user John@domain1.com receives a call from Mary@domain2.com, by default the phone only shows the call as being from Mary, so John is not able to pick up or call back Mary from the received call list. With this parameter set to yes, the phone logs the call as being from Mary@domain2.com, and John can dial Mary from the received call list.

BroadSoft ACD

This parameter enables support for basic BroadSoft Automatic Call Distribution (ACD). The supported values for this option are Yes and No (default).

If you set Broadsoft ACD to Yes, the phone sends a Subscribe message according to the BroadSoft specification.

If you set Broadsoft ACD to No, the phone may still send out a Subscribe message because another feature is using ACD, but the phone ignores any Notify message from the BroadSoft server related to ACD.

Limitations:

• Cisco SPA301 or Cisco SPA501G—ACD is not supported. The ACD Login and Status keys are not visible.
• Cisco SPA509—Lines 9–12 cannot be used as ACD Agents since the Lines cannot be selected for Login/Logout and Agent status.

You can also configure BroadSoft ACD support by adding the following line to your configuration file to configure this feature on line 1:

Auto Ans Page on Active Call

In conjunction with the global Auto Answer Page parameter (found in the Phone tab), this parameter determines the behavior of the phone when a page call arrives.

When set to no, page calls are not auto answered (even if the phone is idle), regardless of the value of the global Auto Answer Page setting in the Phone tab. When the global Auto Answer Page parameter is set to yes, whether page calls are auto answered or not during an active call depends on the per line Auto Answer Page setting (this parameter). Both Auto Answer Page and Auto Answer Page on Active Call are enabled by default.

Feature Key Sync

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

This feature is disabled by default.

Call Park Monitor Enable

When this parameter is set to yes, the phone monitors the call park status against that extension. When the extension is private, the phone sends out the call park SUBSCRIBE message to the server to request call park notification. When the extension is shared, the SCA SUBSCRIBE includes an ACCEPT header with the x-broadworks-callpark-info tag.

By default, a BLF button always sends out a SUBSCRIBE message, and the server notifies the BLF button if there is a call parked against the monitor extension. The phone presents this information to the user without any client-side configuration.

When this parameter is set to no, the primary extension or shared extension will not monitor the call park status.

Enable Broadsoft Hoteling

Default is no.

When this parameter is set to yes, the phone sends out subscription message (without body) to the server; the phone requests “GuestIn” on the registered extension line with input “User ID” and “Password.” The phone then sends out subscription message (with body) to the server with the 'guestAddress' attribute containing the "User ID" and "Password" for authentication.

When this parameter is set to no, the phone sends out unsubscribe message (with empty body) to the server to request Host-Guest Association termination.

Hoteling Subscription Expires

An expiration value is put in the subscription message. Default value is 3600.

Note If this parameter is set to 0, the default 3600 is used instead.

Proxy

SIP proxy server and port number set by the Service Provider for all outbound requests. For example: 192.168.2.100:6060.

Outbound Proxy

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

Alternate Proxy

Alternate Outbound Proxy

These fields are used with the Verizon dual registration/fast fallback feature. This feature provides fast fallback when there is network partition at the Internet or when the primary proxy (or primary outbound proxy) is not responsive or available. The feature works well in a Verizon deployment environment as the alternate proxy is the Integrated Service Router (ISR) with analog outbound phone connection.

Enter the proxy server addresses and port numbers in these fields. After the phone is registered to the primary proxy and the alternate proxy (or primary outbound proxy and alternate outbound proxy), the phone always sends out INVITE and Non-INVITE SIP messages (except registration) via the primary proxy. The phone always registers to both the primary and alternate proxies. If there is no response from the primary proxy after timeout (per the SIP RFC spec) for a new INVITE, the phone attempts to connect with the alternate proxy. The phone always tries the primary proxy first, and immediately tries the alternate proxy if the primary is unreachable.

Active transactions (calls) never fall back between the primary and alternate proxies. If there is fallback for a new INVITE, the subscribe/notify transaction will fall back accordingly so that the phone's state can be maintained properly.

You must also set Dual Registration in the Proxy and Registration section to yes.

Use Outbound Proxy

Enables an outbound proxy (for example, 172.20.2.1:5060—port is optional) or a domain name such as sip.server.com as long as this name is a fully-qualified domain name. If set to no, the Outbound Proxy and Use OB Proxy in Dialog fields are ignored.

Defaults to no.

Optionally, the proxy can be configured (Cisco SPA500 Series only) for Survivable Remote Site Telephony (SRST) support. The proxy is configured with an extension that includes a statically-configured DNS SRV record or DNS A record. Configuring the proxy allows for failover and fallback functionality with a secondary proxy server. For example:

For SRV Record:

Set Use DNS SRV to no and DNS SRV Auto Prefix to no.

For A Record:

Set Use DNS SRV to no and DNS SRV Auto Prefix to no.

Use OB Proxy In Dialog

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

Defaults to yes.

Register

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

Defaults to yes.

Make Call Without Reg

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

Defaults to no.

Register Expires

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

Defaults to 60.

Ans Call Without Reg

If enabled, the user does not have to be registered with the proxy to answer calls.

Defaults to no.

Use DNS SRV

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

Defaults to no.

DNS SRV Auto Prefix

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

Defaults to no.

Proxy Fallback Intvl

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

Defaults to 3600

Proxy Redundancy Method

Select Normal or Based on SRV port. The phone creates an internal list of proxies returned in the DNS SRV records.

If you select Normal, the list contains proxies ranked by weight and priority.

If you select Based on SRV, the phone uses normal, then inspects the port number based on the first listed proxy port.

Defaults to Normal.

Dual Registration

Set to yes to enable the Verizon dual registration/fast fallback feature. You must also configure the alternate proxy/alternate outbound proxy fields in the Proxy and Registration section.

Auto Register when Failover

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

Defaults to no.

There are 2 Failover options:

• Unregister with current proxy, register to backup proxy and then re-send to backup proxy.

This option can be used under the following conditions:

• The parameter Auto Register when Failover is set to yes.
• The failed message should be an INVITE message.
• The return code should match the parameter Try Backup RSC, otherwise the message times out.

Note When the parameter Auto Register when Failover is set to no, if the return code matches with Try Backup RSC or the message times out, the phone re-sends this message directly to backup proxy.

• Re-send directly to backup proxy.

Display Name

Display name for caller ID.

User ID

Extension number for this line.

Password

Password for this line.

Defaults to blank.

Use Auth ID

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

Defaults to no.

Auth ID

Authentication ID for SIP authentication.

Defaults to blank.

Reversed Auth Realm

The Reversed Authentication Realm field. The default value is empty; the proxy address is used as the authentication realm.

To use a different authentication realm, enter the IP address to use in the Reversed Authentication Realm field.

Mini Certificate

Base64 encoded of Mini-Certificate concatenated with the 1024-bit public key of the certificate authority (CA) signing the mini-certificate of all subscribers in the group.

Defaults to blank.

SRTP Private Key

Base64 encoded of the 512-bit private key per subscriber for establishment of a secure call.

Defaults to blank.

Resident Online Number

For Skype use. Enter the Skype Number associated with the registration user ID.

SIP URI

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

"sip:UserName@Domain"

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

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

Preferred Codec

Preferred codec for all calls. (The actual codec used in a call still depends on the outcome of the codec negotiation protocol.) Select one of the following: G711u, G711a, G722, G726-16, G726-24, G726-32, G726-40, G729a, or G723.

Cisco SPA525G or Cisco SPA525G2: G711u, G711a, G726-32, G729a, and G722. G.723 (not available on Cisco SPA300 Series or Cisco SPA500 Series). G722 not available on Cisco WIP310.

Defaults to G711u.

Use Pref Codec Only

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

Defaults to no.

Second Preferred Codec

The second preferred codec when the preferred codec cannot be used. If Use Pref Codec Only is enabled (set to yes), this parameter is not used.

Defaults to Unspecified.

Third Preferred Codec

The third preferred codec when the preferred codec and second preferred codec cannot be used. If Use Pref Codec Only is enabled (set to yes), this parameter is not used.

Defaults to Unspecified.

G711u Enable

Enables use of the G711u codec. Defaults to yes.

G711a Enable

Enables use of the G711a codec. Defaults to yes.

G729a Enable

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

Defaults to yes.

G722 Enable

Enables use of the G.722 codec. Defaults to yes.

Note Not applicable to the Cisco WIP310.

G723 Enable

To enable the use of the G.723a codec at 6.3 kbps, select yes. Otherwise, select no.

Defaults to yes.

Note G.723 is not supported on the Cisco SPA300 Series, Cisco SPA500 Series, or Cisco WIP310.

G726-16 Enable

To enable the use of the G.726 codec at 16 kbps, select yes. Otherwise, select no.

Defaults to yes.

Note Not supported on the Cisco SPA525G or Cisco SPA525G2.

G726-24 Enable

To enable the use of the G.726 codec at 24 kbps, select yes. Otherwise, select no.

Defaults to yes.

Note Not supported on the Cisco SPA525G or Cisco SPA525G2 or Cisco WIP310.

L16 Enable

To enable the use of the L16 codec, select yes. Otherwise, select no.

Defaults to yes.

Note Cisco SPA525G or Cisco SPA525G2 only.

G726-32 Enable

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

Defaults to yes.

G726-40 Enable

To enable the use of the G.726 codec at 40 kbps, select yes. Otherwise, select no.

Defaults to yes.

Note Not applicable to the Cisco SPA525G or Cisco SPA525G2.

Release Unused Codec

Allows the release of codecs not used after codec negotiation on the first call so that other codecs can be used for the second line. To use this feature, select yes.

Defaults to yes.

DTMF Process AVT

Select yes to process RTP DTMF events. Otherwise, select no. If this parameter is set to no, the AVT payload type is not included in outbound SDP.

Defaults to yes.

Silence Supp Enable

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

Defaults to no.

DTMF Tx Method

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

Defaults to Auto.

DTMF Tx Volume for AVT Packet

Allows you to manually configure the AVT Tx volume. The value of this parameter is inserted into the volume field of the payload in the AVT packet.

Values are based on the AVT specification as described in RFC 2833, RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals. According to RFC 2833, the volume field is represented by 6 bits, and describes the power level of the tone, expressed in dBm0 after dropping the sign.

Valid range for this parameter is 0 to 63. If the provisioned value is negative, it will be negated first. Thereafter, if the value is beyond the high limit of 63, it will be clipped to 63.

The default value is 0, and is the recommended setting. However, some gateways do not accept this volume setting. If the gateway does not accept the value of 0, the DTMF tone is not relayed to the remote end. As a workaround for the phone to interoperate with those gateways, you can change the value to a value greater than 0.

Use Remote Pref Codec

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

Codec Negotiation

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

Dial Plan

Dial plan script for this line.

The default is (<9:>xx.)

The dial plan syntax is expanded in the Cisco SPA IP phones to allow the designation of three parameters to be used with a specific gateway:

• uid— authentication user-id
• pwd—authentication password
• nat— if this parameter is present, use NAT mapping

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

Caller ID Map

Inbound caller ID numbers can be mapped to a different string. For example, a number that begins with +44xxxxxx can be mapped to 0xxxxxx. This feature has the same syntax as the Dial Plan parameter. With this parameter, you can specify how to map a caller ID number for display on screen and recorded into call logs. (Not applicable to Cisco WIP310.)

Enable IP Dialing

Enable or disable IP dialing.

Defaults to no.

Cfwd Setting

The overall control to turn on all forward functions. Defaults to yes.

Cfwd All Dest

Enter the extensions to forward calls to.

Cfwd Busy Dest

Enter the extensions to forward calls to when the line is busy.

Defaults to voice mail.

Cfwd No Ans Dest

Enter the extension to forward calls to when the call is not answered.

Defaults to voice mail.

Cfwd No Ans Delay

Enter the time delay in seconds to wait before forwarding a call that is not answered.

Defaults to 20 seconds.

CW Setting

Enables or disables call waiting.

Block CID Setting

Enables or disables the Block Caller ID feature (caller ID is not transmitted).

Block ANC Setting

Enables or disables the blocking of anonymous calls.

DND Setting

Enables or disables Do Not Disturb.

Secure Call Setting

Enables or disables Secure Call.

Dial Assistance