Monitoring Phone Systems

Monitoring Phone Systems Overview

You can view a variety of information about the phone using the phone status menu on the phone and the phone web pages. This information includes:

  • Device information

  • Network setup information

  • Network statistics

  • Device logs

  • Streaming statistics

This chapter describes the information that you can obtain from the phone web page. You can use this information to remotely monitor the operation of a phone and to assist with troubleshooting.

Include a Device Identifier in Uploaded Syslog Messages

You can choose to include a device identifier in syslog messages that are uploaded to the syslog server. While the IP address of a phone may change over time, the device identifier does not change. This can ease the process of identifying the source of each message in a stream of incoming messages from multiple phones. The device identifier appears after the timestamp in each message.

Before you begin

Configure a syslog sever for the phone to upload syslog messages. See Syslog Server in Optional Network Configuration for details.

Procedure


Step 1

On the phone administration web page, go to Voice > System > Optional Network Configuration.

Step 2

Configure the Syslog Identifier parameter as described in Optional Network Configuration.


Cisco IP Phone Status

The following sections describes how to view model information, status messages, and network statistics on the Cisco IP Phone.

  • Model Information: Displays hardware and software information about the phone.

  • Status menu: Provides access to screens that display the status messages, network statistics, and statistics for the current call.

You can use the information that displays on these screens to monitor the operation of a phone and to assist with troubleshooting.

You can also obtain much of this information, and obtain other related information, remotely through the phone web page.

Display the Phone Information Window

Procedure


Step 1

Press Applications .

Step 2

Select Status > Product Information.

When a user password is set, a corresponding icon (lock or certificate) displays at the top-right corner of the phone screen.

Step 3

To exit the Model Information screen, press Back.


View the Phone Status

Procedure


Step 1

Press Applications .

Step 2

Select Status > Phone Status > Phone Status.

You can view the following information:

  • Elapsed time—Total time elapsed since the last reboot of the system

  • Tx (Packets)—Transmitted packets from the phone.

  • Rx (Packets)—Received packets from the phone.


View the Status Messages on the Phone

Procedure


Step 1

Press Applications .

Step 2

Select Status > Status messages.

You can view a log of the various phone statuses since provisioning was last done.

Note 

Status messages reflect UTC time and are not affected by the timezone settings on the phone.

Step 3

Press Back.


View the Network Status

Procedure


Step 1

Press Applications .

Step 2

Select Status > Network Status.

You can view the following information:

  • Network type—Indicates the type of Local Area Netwrok (LAN) connection that the phone uses.

  • Network status—Indicates if the phone is connected to a network.

  • IPv4 status—IP address of the phone. You can see information on IP address, Addressing type, IP status, Subnet mask, Default router, Domain Name Server (DNS) 1, DNS 2 of the phone.

  • IPv6 status —IP address of the phone. You can see information on IP address, Addressing type, IP status, Subnet mask, Default router, Domain Name Server (DNS) 1, DNS 2 of the phone.

  • VLAN ID—VLAN ID of the phone.

  • MAC address—Unique Media Access Control (MAC) address of the phone.

  • Host name—Displays the current host name assigned to the phone.

  • Domain—Displays the network domain name of the phone. Default: cisco.com

  • Switch port link—Status of the switch port.

  • Switch port config—Indicates speed and duplex of the network port.

  • PC port config—Indicates speed and duplex of the PC port.

  • PC port link—Indicates speed and duplex of the PC port.


Display Call Statistics Window

You can access the Call Statistics screen on the phone to display counters, statistics, and voice-quality metrics of the most recent call.


Note

You can also remotely view the call statistics information by using a web browser to access the Streaming Statistics web page. This web page contains additional RTCP statistics that are not available on the phone.


A single call can use multiple voice streams, but data is captured for only the last voice stream. A voice stream is a packet stream between two endpoints. If one endpoint is put on hold, the voice stream stops even though the call is still connected. When the call resumes, a new voice packet stream begins, and the new call data overwrites the former call data.

To display the Call Statistics screen for information about the latest voice stream, follow these steps:

Procedure


Step 1

Press Applications .

Step 2

Select Status > Phone Status > Call Statistics.

Step 3

Press Back.


Call Statistics Fields

The following table describes the items on the Call Statistics screen.

Table 1. Call Statistics Items for the Cisco IP Phone

Item

Description

Receiver Codec

Type of received voice stream (RTP streaming audio from codec):
  • G.729

  • G.722

  • G.711 mu-law

  • G.711 A-law

  • OPUS

  • iLBC

Sender Codec

Type of transmitted voice stream (RTP streaming audio from codec):
  • G.729

  • G.722

  • G.711 mu-law

  • G.711 A-law

  • OPUS

  • iLBC

Receiver Size

Size of voice packets, in milliseconds, in the receiving voice stream (RTP streaming audio).

Sender Size

Size of voice packets, in milliseconds, in the transmitting voice stream.

Rcvr Packets

Number of RTP voice packets that were received since voice stream opened.

Note 

This number is not necessarily identical to the number of RTP voice packets that were received since the call began because the call might have been placed on hold.

Sender Packets

Number of RTP voice packets that were transmitted since voice stream opened.

Note 

This number is not necessarily identical to the number of RTP voice packets that were transmitted since the call began because the call might have been placed on hold.

Avg Jitter

Estimated average RTP packet jitter (dynamic delay that a packet encounters when going through the network), in milliseconds, that was observed since the receiving voice stream opened.

Max Jitter

Maximum jitter, in milliseconds, that was observed since the receiving voice stream opened.

Receiver Discarded

Number of RTP packets in the receiving voice stream that were discarded (bad packets, too late, and so on).

Note 

The phone discards payload type 19 comfort noise packets that Cisco Gateways generate, because they increment this counter.

Rcvr Lost Packets

Missing RTP packets (lost in transit).

Voice-Quality Metrics

Cumulative Conceal Ratio

Total number of concealment frames divided by total number of speech frames that were received from start of the voice stream.

Interval Conceal Ratio

Ratio of concealment frames to speech frames in preceding 3-second interval of active speech. If using voice activity detection (VAD), a longer interval might be required to accumulate 3 seconds of active speech.

Max Conceal Ratio

Highest interval concealment ratio from start of the voice stream.

Conceal Seconds

Number of seconds that have concealment events (lost frames) from the start of the voice stream (includes severely concealed seconds).

Severely Conceal Seconds

Number of seconds that have more than 5 percent concealment events (lost frames) from the start of the voice stream.

Latency

Estimate of the network latency, expressed in milliseconds. Represents a running average of the round-trip delay, measured when RTCP receiver report blocks are received.

View the Customization State in the Configuration Utility

After the RC download from the EDOS server completes, you can view the customization state of a phone using the web interface.

Here are the descriptions of the remote customization states:

  • Open—The phone has booted for the first time and is not configured.

  • Aborted—Remote customization is aborted due to other Provisioning like DHCP options.

  • Pending—The profile has been downloaded from the EDOS server.

  • Custom-Pending—The phone has downloaded a redirect URL from the EDOS server.

  • Acquired—In the profile downloaded from the EDOS server, there is a redirect URL for provision configuration. If the redirect URL download from the provisioning server is successful, this state is displayed.

  • Unavailable—Remote customization has stopped because the EDOS server responded with an empty provisioning file and the HTTP response was 200 OK.

Procedure


Step 1

On the Phone Web page, select Admin Login > Info > Status.

Step 2

In the Product Information section, you can view the customization state of the phone in the Customization field.

If any provisioning is failing, you can view the details in the Provisioning Status section on the same page.


Cisco IP Phone Web Page

This section describes the information that you can obtain from the phone web page. You can use this information to remotely monitor the operation of a phone and to assist with troubleshooting.

Info

The fields on this tab are read-only and cannot be edited.

Status

System Information

Parameter

Description

Host Name

Displays the current host name assigned to the phone.

Domain

Displays the network domain name of the phone.

Default: cisco.com

Primary NTP Server

Displays the primary NTP server assigned to the phone.

Secondary NTP Server

Displays the secondary NTP server assigned to the phone.

IPv4 Information

Parameter

Description

IP Status

Indicates that the connection is established.

Connection Type

Indicates the type of internet connection for the phone:

  • DHCP

  • Static IP

Current IP

Displays the current IP address assigned to the IP phone.

Current Netmask

Displays the network mask assigned to the phone.

Current Gateway

Displays the default router assigned to the phone.

Primary DNS

Displays the primary DNS server assigned to the phone.

Secondary DNS

Displays the secondary DNS server assigned to the phone.

IPv6 Information

Parameter

Description

IP Status

Indicates that the connection is established.

Connection Type

Indicates the type of internet connection for the phone:

  • Static IP

  • DHCP

Current IP

Displays the current IPv6 address assigned to the IP phone.

Prefix Length

Identifies number of bits of a global unicast IPv6 address that are part of`the network. For example, if the IPv6 address is 2001:0DB8:0000:000b::/64, the number 64 identifies that the first 64 bits are part of the network.

Current Gateway

Displays the default router assigned to the phone.

Primary DNS

Displays the primary DNS server assigned to the phone.

Secondary DNS

Displays the secondary DNS server assigned to the phone.

Reboot History

For information about reboot history, see Reboot Reasons.

Product Information

Parameter

Description

Product Name

Model number of the phone.

Software Version

Version number of the phone firmware.

MAC Address

Hardware address of the 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.

Serial Number

Serial number of the phone.

Hardware Version

Version number of the phone hardware.

Client Certificate

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

Downloaded Locale Package

Parameter

Description

Locale download status

Displays the downloaded locale package status.

Locale download URL

Displays the location from where the local package is downloaded.

Font download status

Displays the downloaded font file status.

Font download URL

Displays the location from where the font file is downloaded.

Phone Status

Parameter

Description

Current Time

Current date and time of the system; for example, 08/06/14 1:42:56 a.m.

Elapsed Time

Total time elapsed since the last reboot of the system; for example, 7 days, 02:13:02.

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 which includes retransmissions.

SIP Bytes Recv

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

Network Packets Sent

Total number of network packets sent.

Network Packets Recv

Total number of network packets received.

External IP

External IP of the phone.

Operational VLAN ID

ID of the VLAN currently in use if applicable.

SW Port

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

PC Port

Displays the type of Ethernet connection from PC Port.

Upgrade Status

Displays status of the last phone upgrade.

SW Port Config

Displays the type of SW port configuration.

PC Port Config

Displays the type of PC port configuration.

Last Successful Login

Displays the time when the phone has last successful log in.

Last Failed Login

Displays the time when the phone has last failed log in.

Dot1x Authentication

Parameter

Description

Transaction status

Indicates if the phone is authenticated.

Protocol

Displays the protocol of the registered phone.

Ext Status

Parameter

Description

Registration State

Shows “Registered” if the phone is registered, or “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 Seconds

Number of seconds before the next registration renewal.

Message Waiting

Indicates whether message waiting is enabled or disabled.

Mapped SIP Port

Port number of the SIP port mapped by NAT.

Hoteling State

Indicates whether Hoteling is enabled or disabled.

Extended Function Status

Indicates whether extended function is enabled.

Line Call Status

Parameter

Description

Call State

Status of the call.

Tone

Type of tone that the call uses.

Encoder

Codec used for encoding.

Decoder

Codec used for decoding.

Type

Direction of the call.

Remote Hold

Indicates whether the far end placed the call on hold.

Callback

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

Mapped RTP Port

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

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.

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.

Packets Lost

Number of packets lost.

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).

Packet Discarded

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 that describes 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

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

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

Paging Status

Parameter

Description

Multicast Rx Pkts

Indicates Rx packets during a multicast paging.

Multicast Tx Pkts

Indicates Tx packets during a multicast paging.

TR-069 Status

Parameter

Description

TR-069 Feature

Indicates if TR-069 function is enabled or disabled.

Periodic Inform Time

Displays the inform time interval from CPE to ACS.

Last Inform Time

Indicates the last inform time.

Last Transaction Status

Displays the success or the failure status.

Last Session

Indicates the start and end time of the session.

ParameterKey

Displays the key for reference checkpoint for parameter set configured.

PRT Status

Parameter

Description

PRT Generation Status

The location of initiation and status of generation of the most recently initiated problem report.

Problem reports may be initiated from the phone LCD user interface, from the phone administration web page, or remotely. See Report All Phone Issues from the Phone Web Page and Report a Phone Problem Remotely for details.

XML tag in status.xml: PRT_Generation_Status

PRT Upload Status

The status of upload of the most recently initiated problem report.

See Configure PRT Upload for information on configuring an upload rule for problem reports.

XML tag in status.xml: PRT_Upload_Status

Custom CA Status

These fields display the status of provisioning using a custom Certificate Authority (CA).

Parameter

Description

Custom CA Provisioning Status

Indicates whether provisioning using a custom CA succeeded or failed:

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

  • 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.

Custom CA certificates are configured in the Provisioning tab. For more information about custom CA certificates, see the Cisco IP Phone 7800 Series Multiplatform Phones Provisioning Guide.

Provisioning Status

Parameter

Description

Provisioning Profile

Displays the profile file name of the phone.

Provisioning Status 1

Displays the provisioning status (resync) of the phone.

Provisioning Status 2

Provisioning Status 3

Provisioning Failure Reason

Displays the reason for the failure of provisioning of the phone.


Note

The Upgrade and Provisioning Status are displayed in reverse chronological order (like reboot history). Each entry gives the status, time, and reason.


Debug Info

Console Logs

Displays the syslog output of the phone in the reverse order, where messages is the latest one. The display includes hyperlinks to individual log files. The console log files include debug and error messages received on the phone and the time stamp reflects UTC time, regardless of the time zone settings.

Parameter

Description

Debug Message

Displays debug messages when you click messages link.

Problem Reports

Parameter

Description

Report Problem

Displays the tab Generate PRT.

Prt file

Displays the file name of the PRT logs.

Packet Capture

Displays the tab Start Packet Capture. Click this tab to initiate capture packets. Click All to capture all packets that the phone receives or click Host IP Address to capture packets only when src/dest is the IP address of the phone.

You can also stop the capture process after initiating it.

Capture File

Displays the file that contains the captured packets. Download the file to see the packet details.

Factory Reset

Parameter

Description

Factory Reset

Resets the phone when you click Factory Reset tab when the phone is idle.

Download Status

Firmware Upgrade Status

Parameter

Description

Firmware Upgrade Status 1

Displays the upgrade status (failed or succeeded) with reason for the same.

Firmware Upgrade Status 2

Firmware Upgrade Status 3

Provisioning Status

Parameter

Description

Provisioning Status 1

Displays the provisioning status (resync) of the phone.

Provisioning Status 2

Provisioning Status 3

Custom CA Status

Parameter

Description

Custom CA Provisioning Status

Indicates whether provisioning using a custom CA succeeded or failed:

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

  • 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.

Network Statistics

Ethernet Information

Parameter

Description

TxFrames

Total number of packets that the phone transmitted.

TxBroadcasts

Total number of broadcast packets that the phone transmitted.

TxMulticasts

Total number of multicast packets that the phone transmitted.

TxUnicasts

Total number of unicast packets that the phone transmitted.

RxFrames

Total number of packets received by the phone.

RxBroadcasts

Total number of broadcast packets that the phone received.

RxMulticasts

Total number of multicast packets that the phone received.

RxUnicasts

Total number of unicast packets that the phone received.

Network Port Information

Parameter

Description

RxtotalPkt

Total number of packets that the phone received.

Rxunicast

Total number of unicast packets that the phone received.

Rxbroadcast

Total number of broadcast packets that the phone received.

Rxmulticast

Total number of multicast packets that the phone received.

RxDropPkts

Total number of packets dropped.

RxUndersizePkts

The total number of packets received that are less than 64 octets long, which excludes framing bits, but includes FCS octets, and are otherwise well formed.

RxOversizePkts

The total number of packets received that are longer than 1518 octets, which excludes framing bits, but includes FCS octets, and are otherwise well formed.

RxJabbers

The total number of packets received that are longer than 1518 octets, which excludes framing bits, but incudes FCS octets, and do not end with an even number of octets (alignment error), or had an FCS error.

RxAlignErr

Total number of packets between 64 and 1522 bytes in length that were received and that had a bad Frame Check Sequence (FCS).

Rxsize64

Total number of received packets, including bad packets, that were between 0 and 64 bytes in size.

Rxsize65to127

Total number of received packets, including bad packets, that were between 65 and 127 bytes in size.

Rxsize128to255

Total number of received packets, including bad packets, that were between 128 and 255 bytes in size.

Rxsize256to511

Total number of received packets, including bad packets, that were between 256 and 511 bytes in size.

Rxsize512to1023

Total number of received packets, including bad packets, that were between 512 and 1023 bytes in size.

Rxsize1024to1518

Total number of received packets, including bad packets, that were between 1024 and 1518 bytes in size.

TxtotalGoodPkt

Total number of good packets (multicast, broadcast, and unicast) that the phone received.

lldpFramesOutTotal

Total number of LLDP frames that the phone sent out.

lldpAgeoutsTotal

Total number of LLDP frames that timed out in the cache.

lldpFramesDiscardedTotal

Total number of LLDP frames that were discarded when any of the mandatory TLVs is missing, out of order, or contains out of range string length.

lldpFramesInErrorsTotal

Total number of LLDP frames that were received with one or more detectable errors.

lldpFramesInTotal

Total number of LLDP frames that the phone received.

lldpTLVDiscardedTotal

Total number of LLDP TLVs that were discarded.

lldpTLVUnrecognizedTotal

Total number of LLDP TLVs that were not recognized on the phone.

CDPNeighborDeviceId

Identifier of a device connected to this port that CDP discovered.

CDPNeighborIP

IP address of the neighbor device discovered that CDP discovered.

CDPNeighborPort

Neighbor device port to which the phone is connected discovered by CDP.

LLDPNeighborDeviceId

Identifier of a device connected to this port discovered by LLDP discovered.

LLDPNeighborIP

IP address of the neighbor device that LLDP discovered.

LLDPNeighborPort

Neighbor device port to which the phone connects that LLDP discovered.

PortSpeed

Speed and duplex information.

Access Port Information

Parameter

Description

RxtotalPkt

Total number of packets that the phone received.

Rxunicast

Total number of unicast packets that the phone received.

Rxbroadcast

Total number of broadcast packets that the phone received.

Rxmulticast

Total number of multicast packets that the phone received.

RxDropPkts

Total number of packets dropped.

RxUndersizePkts

The total number of packets received that are less than 64 octets long, which excludes framing bits, but includes FCS octets, and are otherwise well formed.

RxOversizePkts

The total number of packets received that are longer than 1518 octets, which excludes framing bits, but includes FCS octets, and are otherwise well formed.

RxJabbers

The total number of packets received that are longer than 1518 octets, which excludes framing bits, but incudes FCS octets, and do not end with an even number of octets (alignment error), or had an FCS error.

RxAlignErr

Total number of packets between 64 and 1522 bytes in length that were received and that had a bad Frame Check Sequence (FCS).

Rxsize64

Total number of received packets, including bad packets, that were between 0 and 64 bytes in size.

Rxsize65to127

Total number of received packets, including bad packets, that were between 65 and 127 bytes in size.

Rxsize128to255

Total number of received packets, including bad packets, that were between 128 and 255 bytes in size.

Rxsize256to511

Total number of received packets, including bad packets, that were between 256 and 511 bytes in size.

Rxsize512to1023

Total number of received packets, including bad packets, that were between 512 and 1023 bytes in size.

Rxsize1024to1518

Total number of received packets, including bad packets, that were between 1024 and 1518 bytes in size.

TxtotalGoodPkt

Total number of good packets (multicast, broadcast, and unicast) that the phone received.

lldpFramesOutTotal

Total number of LLDP frames that the phone sent out.

lldpAgeoutsTotal

Total number of LLDP frames that timed out in the cache.

lldpFramesDiscardedTotal

Total number of LLDP frames that were discarded when any of the mandatory TLVs is missing, out of order, or contains out of range string length.

lldpFramesInErrorsTotal

Total number of LLDP frames that were received with one or more detectable errors.

lldpFramesInTotal

Total number of LLDP frames that the phone received.

lldpTLVDiscardedTotal

Total number of LLDP TLVs that were discarded.

lldpTLVUnrecognizedTotal

Total number of LLDP TLVs that were not recognized on the phone.

CDPNeighborDeviceId

Identifier of a device connected to this port that CDP discovered.

CDPNeighborIP

IP address of the neighbor device discovered that CDP discovered.

CDPNeighborPort

Neighbor device port to which the phone is connected discovered by CDP.

LLDPNeighborDeviceId

Identifier of a device connected to this port discovered by LLDP discovered.

LLDPNeighborIP

IP address of the neighbor device that LLDP discovered.

LLDPNeighborPort

Neighbor device port to which the phone connects that LLDP discovered.

PortSpeed

Speed and duplex information.

Voice

System

System Configuration

Parameter

Description

Restricted Access Domains

This feature is used when implementing software customization.

Enable Web Server

Enable/disable web server of the IP phone.

Default: Yes

Enable Protocol

Choose the type of protocol:

  • Http

  • Https

If you specify the HTTPS protocol, you must include https: in the URL.

Default: Http

Enable Direct Action Url

Enables the direct action of the URL.

Default: Yes

Session Max Timeout

Allows you to enter maximum timeout of the session.

Default: 3600

Session Idle Timeout

Allows you to enter idle timeout of the session.

Default: 3600

Web Server Port

Allows you to enter port number of the phone web user interface.

Default: 80

  • 80 for protocol HTTP.

  • 443 for protocol HTTPS.

If you specify a port number other than the default value for that protocol, you must include the nondefault port number in the server URL.

Example: https://192.0.2.1:999/admin/advanced

Enable Web Admin Access

Allows you to enable or disable local access to the phone web user interface. Select Yes or No from the drop-down menu.

Default: Yes

Admin Password

Allows you to enter password for the administrator.

Default: Blank

User Password

Allows you to enter password for the user.

Default: Blank

Phone-UI-readonly

Allows you to make the phone menus and options that the phone users see as read-only fields.

Default: No

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.

Default: No

Specific parameters are then designated as "na", "ro", or "rw" using provisioning files. Parameters designated as "na" don't appear on the phone screen. Parameters designated as "ro" aren't editable by the user. Parameters designated as "rw" are editable by the user.

Block Nonproxy SIP

Enables or disables the phone receiving SIP messages from non-proxy server. If you choose Yes, the phone blocks any incoming non-proxy SIP messages except IN-dialog message. If you choose No, the phone does not block any incoming non-proxy SIP messages.

Set Block Nonproxy SIP to No for phones that use TCP or TLS to transport SIP messages. Nonproxy SIP messages transported over TCP or TLS are blocked by default.

Default: No

Network Settings

Parameter

Description

IP Mode

Allows you to select the internet protocol mode in which the phone operates. Options are: IPv4 Only, IPv6 Only, and Dual Mode. In dual mode, the phone can have both IPv4 and IPv6 addresses.

Default: Dual Mode

IPv4 Settings

Parameter

Description

Connection Type

Internet connection type that is configured for the phone. Options are DHCP and Static IP.

Default: DHCP

NetMask

Subnet mask of the phone.

Static IP

IP address of the phone.

Gateway

IP address of the gateway.

Primary DNS

Primary Domain Name Server (DNS) assigned to the phone.

Secondary DNS

Secondary Domain Name Server (DNS) if assigned to the phone.

IPv6 Settings

Parameter

Description

Connection Type

Internet connection type that is configured for the phone. Options are DHCP and Static IP.

Default: DHCP

Static IP

IPv6 address of the phone.

Prefix Length

Identifies number of bits of a global unicast IPv6 address that are part of`the network. For example, if the IPv6 address is 2001:0DB8:0000:000b::/64, the number 64 identifies that the first 64 bits are part of the network.

Gateway

IP address of the gateway.

Primary DNS

Primary Domain Name Server (DNS) assigned to the phone.

Secondary DNS

Secondary Domain Name Server (DNS) if assigned to the phone.

Broadcast Echo

Options are Disabled and Enabled.

Default: Disabled

Auto Config

When enabled, phone generates an IPv6 address by default with the prefix length sent from the router. Options are Disabled and Enabled.

Default: Enabled

802.1X Authentication

Parameter

Description

Enable 802.1X Authentication

Enables/disables 802.1X

Default: No

Optional Network Configuration

Parameter

Description

Host Name

The hostname of the Cisco IP Phone.

Domain

The network domain of the Cisco IP Phone.

If you are using LDAP, see LDAP Configuration.

DNS Server Order

Specifies the method for selecting the DNS server:

  • Manual, DHCP

  • Manual

  • DHCP,Manual

DNS Query Mode

Specified mode of DNS query.

  • Parallel

  • Sequential

DNS Caching Enable

When set to Yes, the DNS query results are not cached.

Default: Yes

Switch Port Config

Allows you to select speed and duplex of the network port. Values are:

  • Auto

  • 10MB half

  • 10MB full

  • 100 MB half

  • 100MB full

  • 100 half

  • 1000 full

PC Port Config

Allows you to select Speed and duplex of the Computer (access) port.

  • Auto

  • 10MB half

  • 10MB full

  • 100 MB half

  • 100MB full

  • 100 half

  • 1000 full

PC PORT Enable

Specifies if PC port is enabled. Options are Yes or No.

Enable PC Port Mirror

Adds the ability to port mirror on the PC port. When enabled, you can see the packets on the phone. Select Yes to enable PC port mirroring and select No to disable it.

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.

Syslog Identifier

Select the device identifier to include in syslog messages that are uploaded to the syslog server. The device identifier appears after the timestamp in each message.

  • None: No device identifier.

  • $MA: The MAC address of the phone, expressed as continuous lower case letters and digits. Example: c4b9cd811e29

  • $MAU: The MAC address of the phone, expressed as continuous upper case letters and digits. Example: C4B9CD811E29

  • $MAC: The MAC address of the phone in the standard colon-separated format. Example: c4:b9:cd:81:1e:29

  • $SN: The product serial number of the phone.

Default: None

Example XML configuration:

<Syslog_Identifier ua="na">$MAC</Syslog_Identifier>

Debug Level

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

Default: 0

Primary NTP Server

IP address or name of the primary NTP server used to synchronize its time.

Default: Blank

Secondary NTP Server

IP address or name of the secondary NTP server used to synchronize its time.

Default: Blank

Enable SSLv3

Choose Yes to enable SSLv3. Choose No to disable.

Default: No

VLAN Settings

Parameter

Description

Enable VLAN

Choose Yes to enable VLAN. Choose No to disable.

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.

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.

PC Port VLAN ID

VLAN ID for the PC port.

DHCP VLAN Option

A predefined DHCP VLAN option to learn the voice VLAN ID. You can use the feature only when no voice VLAN information is available by CDP/LLDP and manual VLAN methods. CDP/LLDP and manual VLAN are all disabled.

Valid values are:

  • Null

  • 128 to 149

  • 151 to 158

  • 161 to 254

Set the value to Null to disable DHCP VLAN option.

Cisco recommends that you use DHCP Option 132.

Inventory Settings

Parameter

Description

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.

SIP

SIP Parameters

Parameter

Description

Max Forward

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

Default: 70

Max Redirection

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

Default: 5

Max Auth

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

Default: 2

SIP User Agent Name

Used in outbound REGISTER requests.

Default: $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.

Default: $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.

Default: 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. 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.

Default: application/dtmf-relay

Hook Flash MIME Type

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

Remove Last Reg

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

Use Compact Header

If set to yes, the phone uses compact SIP headers in outbound SIP messages. If inbound SIP requests contain normal headers, the phone substitutes incoming headers with compact headers. If set to no, the phones use normal SIP headers. If inbound SIP requests contain compact headers, the phones reuse the same compact headers when generating the response, regardless of this setting.

Default: No

Escape Display Name

Enables you to 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.

Default: Yes.

Talk Package

Enables support for the BroadSoft Talk Package that lets users answer or resume a call by clicking a button in an external application.

Default: No

Hold Package

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

Default: No

Conference Package

Enables support for the BroadSoft Conference Package that enables users to start a conference call by clicking a button in an external application.

Default: No

RFC 2543 Call Hold

If set to yes, unit includes 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.

Default: Yes

Random REG CID on Reboot

If set to yes, the 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.

Default: No.

SIP TCP Port Min

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

Default: 5060

SIP TCP Port Max

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

Default: 5080

Caller ID Header

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

Default: PAID-RPID-FROM

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: No

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

If set to yes, it configures the phone to immediately handle NOTIFY sipfrag messages.

Display Diversion Info

Display the Diversion info included in SIP message on LCD or not.

Display Anonymous From Header

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

Sip Accept Encoding

Supports the content-encoding gzip feature. The options are none and gzip.

If gzip is selected, the SIP message header contains the string “Accept-Encoding: gzip”, and the phone is able to process the SIP message body, which is encoded with the gzip format.

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, it disables the display name in “Directory”, “Call History”, and in the “To” header during an outgoing call.

SIP IP Preference

Sets if the phone uses IPv4 or IPv6.

Default: IPv4.

SIP Timer Values (sec)

Parameter

Description

SIP T1

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

Default: 0.5 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.

Default: 4 seconds

SIP T4

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

Default: 5 seconds.

SIP Timer B

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

Default: 16 seconds.

SIP Timer F

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

Default: 16 seconds.

SIP Timer H

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

Default: 16 seconds.

SIP Timer D

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

Default: 16 seconds.

SIP Timer J

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

Default: 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 2000000.

Default: 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 2000000.

Default: 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.

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.

Reg Retry Intv

Interval to wait before the Cisco IP Phone retries registration after failing during the last registration.The range is from 1 to 2147483647

Default: 30

See the note below for additional details.

Reg Retry Long Intvl

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

Default: 1200

See the note below for additional details.

Reg Retry Random Delay

Random delay range (in seconds) to add to <Register Retry Intvl> when retrying REGISTER after a failure. Minimum and maximum random delay to be added to the short timer. The range is from 0 to 2147483647.

Default: 0

Reg Retry Long Random Delay

Random delay range (in seconds) to add to <Register Retry Long Intvl> when retrying REGISTER after a failure.

Default: 0

Reg Retry Intvl Cap

Maximum value of the exponential delay. The maximum value to cap the exponential backoff retry delay (which starts at the Register Retry Intvl and doubles every retry). Defaults to 0, which disables the exponential backoff (that is, the error retry interval is always at the Register Retry Intvl). When this feature is enabled, the Reg Retry Random Delay is added to the exponential backoff delay value. The range is from 0 to 2147483647.

Default: 0

Sub Min Expires

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

Sub Max Expires

Sets the upper limit of the REGISTER minexpires value returned from the Proxy server in the Min-Expires header.

Default: 7200.

Sub Retry Intvl

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

Default: 10.


Note

The phone can use a RETRY-AFTER value when it is received from a SIP proxy server that is too busy to process a request (503 Service Unavailable message). If the response message includes a RETRY-AFTER header, the phone waits for the specified length of time before to REGISTER again. If a RETRY-AFTER header is not present, the phone waits for the value specified in the Reg Retry Interval or the Reg Retry Long Interval.


Response Status Code Handling

Parameter

Description

Try Backup RSC

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

Default: Blank

For example, you can enter numeric values 500 or a combination of numeric values plus wild cards if multiple values are possible. For the later, you can use 5?? to represent all SIP Response messages within the 500 range. If you want to use multiple ranges, you can add a comma "," to delimit values of 5?? and 6??

Retry Reg RSC

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

Default: Blank

For example, you can enter numeric values 500 or a combination of numeric values plus wild cards if multiple values are possible. For the later, you can use 5?? to represent all SIP Response messages within the 500 range. If you want to use multiple ranges, you can add a comma "," to delimit values of 5?? and 6??

RTP Parameters

Parameter

Description

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 16538.

Default: 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 16538.

The maximum value for the RTP port must be lesser than 49152.

Default: 16538

RTP Packet Size

Packet size in seconds, which can range from 0.01 to 0.13. Valid values must be a multiple of 0.01 seconds.

Default: 0.02

Max RTP ICMP Err

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

RTCP Tx Interval

Interval for sending out RTCP sender reports on an active connection. It can range from 0 to 255 seconds.

Default: 0

SDP IP Preferences

Select IPv4 or IPv6.

Default: IPv4

If the phone is in dual-mode and has both ipv4 and ipv6 addresses, it will always include both addresses in SDP by attributes "a=altc …

If IPv4 address is selected, then ipv4 address has higher priority than ipv6 address in SDP and indicates that phone prefers using ipv4 RTP address.

If the phone has only ipv4 address or ipv6 address, SDP does not have ALTC attributes and RTP address is specified in “c=” line.

SDP Payload Types

Parameter

Description

G722.2 Dynamic Payload

G722 Dynamic Payload type.

Default: 96

iLBC Dynamic Payload

iLBC Dynamic Payload type.

Default: 97

iSAC Dynamic Payload

iSAC Dynamic Payload type.

Default: 98

OPUS Dynamic Payload

OPUS Dynamic Payload type.

Default: 99

AVT Dynamic Payload

AVT dynamic payload type. Ranges from 96-127.

Default: 101

INFOREQ Dynamic Payload

INFOREQ Dynamic Payload type.

H264 BP0 Dynamic Payload

H264 BPO Dynamic Payload type.

Default: 110

H264 HP Dynamic Payload

H264 HP Dynamic Payload type.

Default: 110

G711u Codec Name

G711u codec name used in SDP.

Default: PCMU

G711a Codec Name

G711a codec name used in SDP.

Default: PCMA

G729a Codec Name

G729a codec name used in SDP.

Default: G729a

G729b Codec Name

G729b codec name used in SDP.

Default: G729b

G722 Codec Name

G722 codec name used in SDP.

Default: G722

G722.2 Codec Name

G722.2 codec name used in SDP.

Default: G722.2

iLBC Codec Name

iLBC codec name used in SDP.

Default: iLBC

iSAC Codec Name

iSAC codec name used in SDP.

Default: iSAC

OPUS Codec Name

OPUS codec name used in SDP.

Default: OPUS

AVT Codec Name

AVT codec name used in SDP.

Default: telephone-event

NAT Support Parameters

Parameter

Description

Handle VIA received

Enables the phone to process the received parameter in the VIA header.

Default: No

Handle VIA rport

Enables the phone to process the rport parameter in the VIA header.

Default: No

Insert VIA received

Enables to insert the received parameter into the VIA header of SIP responses if the received-from IP and VIA sent-by IP values differ.

Default: No

Insert VIA rport

Enables to insert the rport parameter into the VIA header of SIP responses if the received-from IP and VIA sent-by IP values differ.

Default: No

Substitute VIA Addr

Enables the user to use NAT-mapped IP:port values in the VIA header.

Default: No

Send Resp To Src Port

Enables to send responses to the request source port instead of the VIA sent-by port.

Default: No

STUN Enable

Enables the use of STUN to discover NAT mapping.

Default: No

STUN Test Enable

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

Default: 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.

Default: Blank

EXT IP

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

If this parameter is specified, phone assumes this IP address when generating SIP messages and SDP (if NAT Mapping is enabled for that line).

Default: Blank

EXT RTP Port Min

External port mapping number of the RTP Port Minimum 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.

Default: 0

NAT Keep Alive Intvl

Interval between NAT-mapping keep alive messages.

Default: 15

Redirect Keep Alive

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

Provisioning

Configuration Profile

Parameter

Description

Provision Enable

Allows or denies resync actions.

Default: 160,159,66,150

Resync On Reset

The device performs a resync operation after power-up and after each upgrade attempt when set to Yes.

Default: Yes

Resync Random Delay

A random delay following the boot-up sequence before performing the reset, specified in seconds. In a pool of IP Telephony devices that are scheduled to simultaneously power up, this introduces a spread in the times at which each unit sends a resync request to the provisioning server. This feature can be useful in a large residential deployment, in the case of a regional power failure.

The value for this field must be an integer ranging between 0 and 65535.

The default value is 2.

Resync At (HHmm)

The time (HHmm) that the device resynchronizes with the provisioning server.

The value for this field must be a four-digit number ranging from 0000 to 2400 to indicate the time in HHmm format. For example, 0959 indicates 09:59.

The default value is empty. If the value is invalid, the parameter is ignored. If this parameter is set with a valid value, the Resync Periodic parameter is ignored.

Resync At Random Delay

Prevents an overload of the provisioning server when a large number of devices power-on simultaneously.

To avoid flooding resync requests to the server from multiple phones, the phone resynchronizes in the range between the hours and minutes, and the hours and minutes plus the random delay (hhmm, hhmm+random_delay). For example, if the random delay = (Resync At Random Delay + 30)/60 minutes, the input value in seconds is converted to minutes, rounding up to the next minute to calculate the final random_delay interval.

The valid value ranges between 0 and 65535.

This feature is disabled when this parameter is set to zero. The default value is 600 seconds (10 minutes).

Resync Periodic

The time interval between periodic resynchronizes with the provisioning server. The associated resync timer is active only after the first successful sync with the server.

The valid formats are as follows:

  • An integer

    Example: An input of 3000 indicates that the next resync occurs in 3000 seconds.

  • Multiple integers

    Example: An input of 600,1200,300 indicates that the first resync occurs in 600 seconds, the second resync occurs in 1200 seconds after the first one, and the third resync occurs in 300 seconds after the second one.

  • A time range

    Example, an input of 2400+30 indicates that the next resync occurs in between 2400 and 2430 seconds after a successful resync.

Set this parameter to zero to disable periodic resynchronization.

The default value is 3600 seconds.

Resync Error Retry Delay

If a resync operation fails because the IP Telephony device was unable to retrieve a profile from the server, or the downloaded file is corrupt, or an internal error occurs, the device tries to resync again after a time specified in seconds.

The valid formats are as follows:

  • An integer

    Example: An input of 300 indicates that the next retry for resync occurs in 300 seconds.

  • Multiple integers

    Example: An input of 600,1200,300 indicates that the first retry occurs in 600 seconds after the failure, the second retry occurs in 1200 seconds after the failure of the first retry, and the third retry occurs in 300 seconds after the failure of the second retry.

  • A time range

    Example, an input of 2400+30 indicates that the next retry occurs in between 2400 and 2430 seconds after a resync failure.

If the delay is set to 0, the device does not try to resync again following a failed resync attempt.

Forced Resync Delay

Maximum delay (in seconds) the phone waits before performing a resynchronization.

The device does not resync while one of its phone lines is active. Because a resync can take several seconds, it is desirable to wait until the device has been idle for an extended period before resynchronizing. This allows a user to make calls in succession without interruption.

The device has a timer that begins counting down when all of its lines become idle. This parameter is the initial value of the counter. Resync events are delayed until this counter decrements to zero.

The valid value ranges between 0 and 65535.

The default value is 14,400 seconds.

Resync From SIP

Controls requests for resync operations via a SIP NOTIFY event sent from the service provider proxy server to the IP Telephony device. If enabled, the proxy can request a resync by sending a SIP NOTIFY message containing the Event: resync header to the device.

Default: Yes

Resync After Upgrade Attempt

Enables or disables the resync operation after any upgrade occurs. If Yes is selected, sync is triggered.

Default: Yes

Resync Trigger 1

Resync Trigger 2

If the logical equation in these parameters evaluates to FALSE, Resync is not triggered even when Resync On Reset is set to TRUE. Only Resync via direct action URL and SIP notify ignores these Resync Trigger.

Default: Blank

Resync Fails On FNF

A resync is considered unsuccessful if a requested profile is not received from the server. This can be overridden by this parameter. When it is set to No, the device accepts a file-not-found response from the server as a successful resync.

Default: Yes

Profile Authentication Type

Specifies the credentials to use for profile account authentication. The available options are:

  • Disabled: Disables the profile account feature. When this feature is disabled, the Profile account setup menu doesn't display on the phone screen.

  • Basic HTTP Authentication: The HTTP login credentials are used to authenticate the profile account.

  • XSI Authentication: XSI login credentials or XSI SIP credentials are used to authenticate the profile account. The authentication credentials depend on the XSI Authentication Type for the phone:

    • When the XSI Authentication Type for the phone is set to Login Credentials, the XSI login credentials are used.

    • When the XSI Authentication Type for the phone is set to SIP Credentials, the XSI SIP credentials are used.

Default: Basic HTTP Authentication

Profile Rule

Profile Rule B

Profile Rule C

Profile Rule D

Each profile rule informs the phone of a source from which to obtain a profile (configuration file). During every resync operation, the phone applies all the profiles in sequence.

Default: /$PSN.xml

If you are applying AES-256-CBC encryption to the configuration files, specify the encryption key with the --key keyword as follows:

[--key <encryption key>]

You can enclose the encryption key in double-quotes (") optionally.

DHCP Option To Use

DHCP options, delimited by commas, used to retrieve firmware and profiles.

Default: 66,160,159,150,60,43,125

DHCPv6 Option To Use

DHCP options, delimited by commas, used to retrieve firmware and profiles.

Default: 17,160,159

Log Request Msg

The message sent to the syslog server at the start of a resync attempt.

Default:

$PN $MAC –Requesting % $SCHEME://$SERVIP:$PORT$PATH

Log Success Msg

The syslog message issued upon successful completion of a resync attempt.

Default:

$PN $MAC -Successful Resync  % 
$SCHEME://$SERVIP:$PORT$PATH 

Log Failure Msg

The syslog message that is issued after a failed download attempt.

Default:
$PN $MAC -- Resync failed: $ERR

User Configurable Resync

Allows a user to resync the phone from the phone screen.

Default: Yes

Upload Configuration Options

Field Description
Report Rule

Specifies how the phone reports its current internal configuration to the provisioning server. The URLs in this field specify the destination for a report and can include an encryption key.

You can use the following keywords, encryption key, and file locations and names to control how you store the phone configuration information:

  • No keywords and only an XML file reports the entire configuration data to server.

  • [--status] keyword reports the status data to server.

  • [--delta] keyword reports the changed configuration to server.

  • [--key <encryption key>] keyword tells the phone to apply AES-256-CBC encryption with the specified encryption key to the configuration report, before sending it to the server.

    You can enclose the encryption key in double-quotes (") optionally.

    Note 
    If you have provisioned the phone with Input Keying Material (IKM) and want the phone to apply RFC 8188-based encryption to the file, do not specify a AES-256-CBC encryption key.
  • Two rules used together as:

    [--delta]http://my_http_server/config-mpp-delta.xml [--status]http://my_http_server/config-mpp-status.xml
Caution 
If you need to use the [--delta]xml-delta file rule and the [--status]xml-status file rule together, you must separate the two rules with a space.

HTTP Report method:

Specifies whether the HTTP Request that the phone sends should be an HTTP PUT or an HTTP POST.

  • PUT Method–To create a new report or overwrite an existing report at a known location on the server. For example, you may want to keep overwriting each report that you send and only store the most current configuration on the server.

  • POST Method–To send the report data to the server for processing, such as, by a PHP script. This approach provides more flexibility for storing the configuration information. For example, you may want to send a series of phone status reports and store all the reports on the server.

Report to Server:

Defines when the phone reports its configuration to the provisioning servers.

  • On Request: The phone reports its configuration only when an administrator sends a sip notify event, or the phone restarts.

  • On Local Change: The phone reports its configuration when any configuration parameter changes by an action on the phone or on the phone administration web page. The phone waits for a few seconds after a change is made, and then reports the configuration. This delay ensures that changes are reported to the web server in batches, rather than reporting a single change at a time.

  • Periodically: The phone reports its configuration at regular intervals. The interval is expressed in seconds.

Example XML configuration:

<Report_to_Server ua="na">

Periodically

</Report_to_Server>
Periodic Upload to Server:

Defines the interval (in seconds) that the phone reports its configuration to the provisioning servers.

This field is used only when Report to Server is set to Periodically.

Default: 3600

Minimum: 600

Maximum: 2592000 (30 days)

Example XML configuration:

<Report_to_Server ua="na">

Periodically

</Report_to_Server>

<!available options: On Request | On Local Change| Periodically-->

<periodic_upload_to_server ua="na">

3600

</periodic_upload_to_server><User_Configurable_Resync ua="na">

Yes

</User_Configurable_Resync_

Upload Delay On Local Change:

Defines the delay (in seconds) that the phone waits after a change is made, and then reports the configuration.

This field is used only when Report to Server is set to On Local Change.

Default: 60

Minimum: 10

Maximum: 900

Example XML configuration:

<Upload_Delay_On_Local_Change ua="na">

60

</Upload_Delay_On_Local_Change>

Firmware Upgrade

Parameter

Description

Upgrade Enable

Allows firmware update operations independent of resync actions.

Default: Yes

Upgrade Rule

A firmware upgrade script that defines upgrade conditions and associated firmware URLs. It uses the same syntax as Profile Rule.

Use the following format to enter the upgrade rule:

protocol://server[:port]/profile_pathname

For example:

tftp://192.168.1.5/image/sip88xx.11-1-1MPP-221.loads

If no protocol is specified, TFTP is assumed. If no server-name is specified, the host that requests the URL is used as the server name. If no port is specified, the default port is used (69 for TFTP, 80 for HTTP, or 443 for HTTPS).

You can also include the credentials that are used to access the server. Then, the upgrade rule is:

[--uid $userID --pwd $password]protocol://server[:port]/profile_pathname

For example,

[--uid TEST --pwd TestAbC123]tftp://192.168.1.5/image/sip88xx.11-1-1MPP-221.loads

If the user ID or the password contains special characters (/ [ & } (*) # , etc.), you need to quote them in the upgrade rule. There are two options for quoting special characters:

  • Put the user ID or the password that contains special characters into double quotation marks (" "). This option doesn't work for some of the special characters, such as " " [ ].

    For example,

    [--uid TEST --pwd "Test#\AbC123"]tftp://192.168.1.5/image/sip88xx.11-1-1MPP-221.loads

  • Use the octal encoding of the special characters.

    For example, escape the pond (#) with "\043" and the backslash with "\057" for the password "Test#\AbC123" in the following rule:

    [--uid TEST --pwd Test\043\057AbC123]tftp://192.168.1.5/image/sip88xx.11-1-1MPP-221.loads

Default: Blank

Log Upgrade Request Msg

Syslog message issued at the start of a firmware upgrade attempt.

Default: $PN $MAC -- Requesting upgrade $SCHEME://$SERVIP:$PORT$PATH

Log Upgrade Success Msg

Syslog message issued after a firmware upgrade attempt completes successfully.

Default: $PN $MAC -- Successful upgrade $SCHEME://$SERVIP:$PORT$PATH -- $ERR

Log Upgrade Failure Msg

Syslog message issued after a failed firmware upgrade attempt.

Default: $PN $MAC -- Upgrade failed: $ERR

Peer Firmware Sharing

Enables or disables the Peer Firmware Sharing feature. Select Yes or No to enable or to disable the feature.

Default: Yes

Peer Firmware Sharing Log Server

Indicates the IP address and the port to which the UDP message is sent.

For example: 10.98.76.123:514 where, 10.98.76.123 is the IP address and 514 is the port number.

For more information about the Provisioning page, see the Cisco IP Phone 7800 Series Multiplatform Phones Provisioning Guide.

CA Settings

Parameter

Description

Custom CA Rule

The URL to download Custom CA.

Default: Blank

HTTP Settings

Parameter

Description

HTTP User Agent Name

Allows you to enter a name for HTTP user.

Default: Blank

Problem Report Tool

Parameter

Description

PRT Upload Rule

Specifies the path to the PRT upload script. You can enter the path in the format:

https://proxy.example.com/prt_upload.php

or

http://proxy.example.com/prt_upload.php

If PRT Max Timer and PRT Upload Rule fields are empty, problem reports are not generated.

PRT Upload Method

Determines the method used to upload PRT logs to the remote server. Options are: HTTP POST and PUT.

Default: POST

PRT Max Timer

Determines at what interval (minutes) the phone starts generating problem report automatically. The interval range that you can set is 15 minutes to 1440 minutes.

Default: Empty

If PRT Max Timer and PRT Upload Rule fields are empty, problem reports are not generated.

a

PRT Name

Defines a name for the generated PRT file. Enter the name in the format:

prt-string1-$MACRO

General Purpose Parameters

Parameter

Description

GPP A - GPP P

The general purpose parameters GPP_* are used as free string, registers when configuring the Cisco IP phones to interact with a particular provisioning server solution. They can be configured to contain diverse values, including the following:

  • Encryption keys

  • URLs

  • Multistage provisioning status information

  • Post request templates

  • Parameter name alias maps

  • Partial string values, eventually combined into complete parameter values

Default: Blank

Regional

Call Progress Tones

Parameter

Description

Dial Tone

Prompts the user to enter a phone number.

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.

Prompt Tone

Prompts the user to enter a call forwarding phone number.

Busy Tone

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

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.

Off Hook Warning Tone

Played when the phone receiver has been off hook after a period of time.

Ring Back Tone

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

Call Waiting Tone

Played when a call is waiting.

Confirm Tone

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

MWI Dial Tone

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

Cfwd Dial Tone

Played when all calls are forwarded.

Holding Tone

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

Conference Tone

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

Secure Call Indication Tone

Played when a call has been successfully switched tosecure 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.

Page Tone

Specifies the tone transmitted when the paging feature is enabled.

Alert Tone

Played when an alert occurs.

Mute Tone

Played when the Mute button is pressed to mute the phone.

Unmute Tone

Played when the Mute button is pressed to unmute the phone.

System Beep

Audible notification tone played when a system error occurs.

Call Pickup Tone

Provides the ability to configure an audio indication for call pickup.

Distinctive Ring Patterns

Parameter

Description

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).

Control Timer Values (sec)

Parameter

Description

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.

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.

Default: 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.

Default: 3

Vertical Service Activation Codes

Parameter

Description

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 *88.

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

Removes caller ID blocking on the next inbound 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 a secure call.

Default: *18.

Secure One Call Deact Code

Disables secure call feature.

Default: *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.

Referral Services Codes

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

One or more *code can be configured into this parameter, such as *98, or *97|*98|*123, and so on. 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 phone to perform a blind transfer to a target number that is prepended by the service *code.

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

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

Feature Dial Services Codes

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

One or more *code can be configured into this parameter, such as *72, or *72|*74|*67|*82, 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 phone to call the target number prepended by the *code. For example, after user dials *72, the phone plays a prompt tone awaiting the user to enter a valid target number. When a complete number is entered, the phone sends a INVITE to *72<target_number> as in a normal call. This feature allows the proxy to process features like call forward (*72) or BLock Caller ID (*67).

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

You can add a parameter to each *code in Features Dial Services Codes to indicate what tone to play after the *code is entered, such as *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 phone plays Prompt tone by default.

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

Vertical Service Announcement Codes

Parameter

Description

Service Annc Base Number

Defaults to blank.

Service Annc Extension Codes

Defaults to blank.

Outbound Call Codec Selection Codes

Parameter

Description

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.

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.

Prefer G722.2 Code

Makes this codec the preferred codec for the associated call.

Force G722.2 Code

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

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.

Prefer iLBC Code

Makes this codec the preferred codec for the associated call.

Force iLBC Code

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

Prefer ISAC Code

Makes this codec the preferred codec for the associated call.

Force ISAC Code

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

Prefer OPUS Code

Makes this codec the preferred codec for the associated call.

Force OPUS Code

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

Time

Parameter

Description

Set Local Date (mm/dd/yyyy)

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

Default: Blank

Set Local Time (HH/mm)

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

Default: Blank

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.

Default: GMT-08:00

Time Offset (HH/mm)

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

Default: 00/00

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.

Default: Yes.

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).

Default: 3/-1/7/2;end=10/-1/7/2;save=1.

Daylight Saving Time Enable

Enables Daylight Saving Time.

Default: Yes

Language

Parameter

Description

Dictionary Server Script

Use this field to specify the language options for the phone display, and the dictionary and font files required for each language. See Set Up Dictionaries and Fonts.

Default: Blank

Language Selection

Use this field to specify the default language. The value must match one of the languages supported by the dictionary server. See Specify a Language for the Phone Display.

You can configure the language through the XML Configuration file. For example:


<Language_Selection ua="na"> Spanish
</Language_Selection>

The language name can have up to 512 characters.

Locale

Use this drop-down list box to see the supported languages. See Supported Languages for the Phone Display.

Phone

General

Parameter

Description

Station Name

Name of the phone.

Station Display Name

Name to identify the phone; appears on the phone screen. You can use spaces in this field and the name does not have to be unique.

Voice Mail Number

A phone number or URL to check voice mail.

Default: None

Handsfree

Parameter

Description

Bluetooth Mode

Shows the method of Bluetooth connection.

  • Phone—Pairs with a Bluetooth headset only.

  • Handsfree—Operates as a handsfree device with a Bluetooth-enabled mobile phone.

  • Both—Uses a Bluetooth headset, or operates with a Bluetooth-enabled mobile phone.

Line

Specifies the line number for which the Bluetooth is enabled.

Line Key

Each line key has a set of settings.

Parameter

Description

Extension

Specifies the n extension to be assigned to Line Key n.

Default: n

XML configuration examples:

To set the line key 1 to extension 1:

<Extension_1_ ua="na">1</Extension_1_>

To disable the extension function for line key 2:

<Extension_2_ ua="na">Disabled</Extension_2_>

Short Name

Specifies the user name for Line Key.

Default: $USER

Share Call Appearance

Specifies whether the incoming call appearance is shared with other phones or it is private.

Extended Function

Use to assign any of the following features or functions to unused line keys on the phone:

  • Busy Lamp Field

  • Call Pickup

  • Speed Dial

Miscellaneous Line Key Settings

Parameter

Description

Line ID Mapping

Specifies the shared call appearance line ID mapping. If Vertical First is set, the second call makes the next available line ID LED flash. If Horizontal first is set, the second call will make the same LED flash on which the first call is received. Also, the behavior is same for both outgoing and incoming calls.

Note 

7811 Cisco IP Phone does not support Line ID Mapping.

Default: Horizontal First

SCA Barge-In Enable

Enables the SCA Barge-In.

Default: No

SCA Sticky Auto Line Seize

If enabled, restricts to automatically pick up an incoming call on a shared line when you take the phone off-hook.

Call Appearances Per Line

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

Default: 2

Supplementary Services

Parameter

Description

Conference Serv

Enable or disable three-way conference service.

Default: Yes

Attn Transfer Serv

Enable or disable attended-call-transfer service.

Default: Yes

Blind Transfer Serv

Enable or disable blind-call-transfer service.

Default: Yes

DND Serv

Enable or disable do not disturb service.

Default: Yes

Block ANC Serv

Enable or disable block-anonymous-call service.

Default: Yes

Block CID Serv

Enable or disable blocking outbound Caller-ID service.

Default: Yes

Secure Call Serv

Enable or disable secured call services.

Default: Yes

Cfwd All Serv

Enable or disable call-forward-all service.

Default: Yes

Cfwd Busy Serv

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

Default: Yes

Cfwd No Ans Serv

Enable or disable call-forward-no-answer service.

Default: Yes

Paging Serv

Enable or disable paging service on the phone.

Default: Yes

Call Park Serv

Enable or disable call park services on the phone.

Default: Yes

Call Pick Up Serv

Enable or disable call pick up services on the phone.

Default: Yes

ACD Login Serv

Enable or disable ACD login services on the phone.

Default: Yes

Group Call Pick Up Serv

Enable or disable group call pick up services on the phone.

Default: Yes

Service Annc Serv

Enable or disable the vertical service announcement services on the phone.

Default: No

Call Recording Serv

Enable or disable the call recording services on the phone.

Default: No

Reverse Phone Lookup Serv

Enable or disable reverse name lookup for the phone.

When enabled, the phone can searches the personal address book and call history, server directory, and either the configured LDAP or XML directory.

Default: Yes

Ringtone

Parameter

Description

Ring1 to Ring12

Ring tone scripts for different rings.

Silent Ring Duration

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.

Extension Mobility

Parameter

Description

EM Enable

Options to enable or to disable the extension mobility support for the phone.

Default: No

EM User Domain

Name of the domain for the phone or the authentication server.

Default: Blank

Session Timer(m)

Specifies the duration of the phone session.

Countdown Timer(s)

Specifies the duration for which it waits before it logs out.

Default: 10

Preferred Password Input Mode

Options to specify the password input method of extension mobility PIN. Options are: Alpha-numeric and Numeric.

Default: Alphanumeric

XSI Phone Service

Parameter

Description

XSI Host Server

Enter the name of the server; for example, xsi.iop1.broadworks.net.

Note 

XSI Host Server uses http protocol by default. To enable XSI over HTTPS, you can specify https:// in the server.

Default: Blank

XSI Authentication Type

Determines the XSI authentication type. Select Login Credentials to authenticate access with XSI id and password. Select SIP Credentials to authenticate access with the register user ID and password of the SIP account registered on the phone.

Default: Login Credentials

Login User ID

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

Enter SIP Auth ID when you select Login Credentials or SIP Credentials for XSI authentication type.

When you choose SIP Auth ID as SIP Credentials, you must enter Login User ID. Without Login User ID, the BroadSoft directory will not appear under the phone Directory list.

Default: Blank

Login Password

Alphanumeric password associated with the User ID.

Enter login password, when you select Login Credentials for XSI authentication type.

Default: Blank

SIP Auth ID

The registered user ID of the SIP account registered on the phone.

Enter SIP Auth ID when you select SIP Credentials for XSI authentication type.

SIP Password

The password of the SIP account registered on the phone.

Enter SIP password when you select SIP Credentials for XSI authentication type.

Directory Enable

Enables BroadSoft directory for the phone user. Select Yes to enable the directory and select No to disable it.

Default: No

Directory Name

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

Default: Blank

Directory Type

Select the type of BroadSoft directory:

Enterprise: 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.

Default: Enterprise

CallLog Enable

Enables to log XSI calls. Select Yes to log XSI calls and select No to disable it.

Default: No

CallLog Associated Line

Allows you to select a phone line for which you want to display the recent call logs.

You can select line numbers ranges from 1 to 10.

Display Recents From

Allows you to set which type of recent call logs the phone will display. Choose Server to display BroadSoft XSI recent call logs and select Phone to display local recent call logs.

Note 

The Display recents from is added to the Recents screen of the phone only when you set CallLog Enable to Yes and Display Recents From type to Server.

Broadsoft XMPP

Parameter

Description

XMPP Enable

Set to Yes to enable the BroadSoft XMPP directory for the phone user.

Default: No

Server

Enter the name of the XMPP server; for example, xsi.iop1.broadworks.net.

Default: Blank

Port

Server port for the directory.

Default: Blank

User ID

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

Default: Blank

Password

Alphanumeric password associated with the User ID.

Default: Blank

Login Invisible

When enabled, the user's presence information is not published when the user signs in.

Default: No

Retry Intvl

Interval, in seconds, to allow a reconnect without a log in after the client disconnects from the server. After this interval, the client needs to reauthenticate.

Default: 30

XML Service

Parameter

Description

XML Directory Service Name

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

Default: Blank

XML Directory Service URL

URL where the XML Directory is located.

Default: Blank

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

Default: Blank

XML Password

XML service password for authentication purposes

Default: Blank

CISCO XML EXE Enable

Enables or disables Cisco XML EXE authentication.

Default: No

CISCO XML EXE Auth Mode

Specifies the authentication mode for Cisco XML EXE. The available options are:

  • Trusted—No authentication is performed (local user password is set or not).

  • Local Credential—Authentication is based on digest authentication using the local user password, if the local user password is set. If not set, then no authentication is performed.

  • Remote Credential—Authentication is based on digest authentication using the remote username/password as set in the XML application on the web page (to access an XML application server).

Default: Trusted

Multiple Paging Group Parameters

Feature

Description

Group Paging Script

Enter a string to configure group paging and priority paging (out of band paging) that doesn’t require the phone registration.

LDAP

Parameter

Description

LDAP Dir Enable

Choose Yes to enable LDAP.

Default: No

Corp Dir Name

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

Default: Blank

Server

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

nnn.nnn.nnn.nnn

Enter the host name of the LDAP server if the MD5 authentication method is used.

Default: Blank

Search Base

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

dc=cv2bu,dc=com

Default: Blank

Client DN

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

dc=cv2bu,dc=com

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

cn=”David Lee”,dc=users,dc=cv2bu,dc=com

cn=”David Lee”,dc=cv2bu,dc=com

username@domain is the client DN format for a Windows server

For example, DavidLee@cv2bu.com

Default: Blank

User Name

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

Default: Blank

Password

Enter the password for the LDAP username.

Default: Blank

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.

Default: None

Last Name Filter

Use this field to specify how the phone must perform searches based on the last name or surname (sn), when users search for contacts.

Examples:

sn:(sn=$VALUE*) instructs the phone to find all last names that begin with the entered search string.

sn:(sn=*$VALUE*) instructs the phone to find all last names in which the entered search string appears anywhere in the last name. This method is more inclusive and retrieves more search results. This method is consistent with the search method in other directories such as the Broadsoft directories and the user's personal address book on the phone.

Default: Blank

First Name Filter

Use this field to specify how the phone must perform searches based on the first name or common name (cn), when users search for contacts.

Examples:

cn:(cn=$VALUE*) instructs the phone to find all first names that begin with the entered search string.

cn:(cn=*$VALUE*) instructs the phone to find all first names in which the entered search string appears anywhere in the first name. This method is more inclusive and retrieves more search results. This method is consistent with the search method in other directories such as the Broadsoft directories and the user's personal address book on the phone.

Default: Blank

Search Item 3

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

Default: Blank

Search Item 3 Filter

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

Default: Blank

Search Item 4

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

Default: Blank

Search Item 4 Filter

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

Default: Blank

Display Attrs

Format of LDAP results displayed 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, 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 is ignored.

  • p—phone number

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

For example, a=givenName,n=firstname;a=sn,n=lastname;a=cn,n=cn;a=telephoneNumber,n=tele,t=p

Default: Blank

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.

Default: Blank

Programmable Softkeys

Parameter

Description

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 there is a missed call.

Off Hook Key List

Softkeys that display when the phone is off-hook.

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.

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 released.

Hold Key List

Softkeys that display when one or more calls are on hold.

Ringing Key List

Softkeys that display when a call is incoming.

To silence an incoming call, you can add Ignore softkey.

Shared Active Key List

Softkeys that display when a call is active on a shared line.

Shared Held Key List

Softkeys that display when a call is on hold on a shared line.

PSK 1 through PSK 16

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.

Extension

General

Parameter

Description

Line Enable

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

Default: Yes

Example XML configuration:

To disable service on the line associated with extension 2:

<Line_Enable_2_ ua="na">No</Line_Enable_2_>

Video Configuration

Parameter

Description

H264 BP0 Enable

Enables the H264 Base Profile 0 codec when you select Yes and disables it when you select No.

Default: Yes

H264 HP Enable

Enables the H264 High Profile codec when you select Yes and disables it when you select No.

Default: Yes

Encryption Method

Selects the encryption method to be used during a secured call. Options are AES 128 and AES 256 GCM.

Default: AES 128

Share Line Appearance

Parameter

Description

Share Ext

Indicates whether this extension is to be shared with other Cisco IP phones or private.

Default: Yes

Shared User ID

The user identified assigned to the shared line appearance.

Default: Blank

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.

Default: 3600

Restrict MWI

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

Default: No

NAT Settings

Parameter

Description

NAT Mapping Enable

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

Default: No

NAT Keep Alive Enable

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

Default: 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.

Default: $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.

Network Settings

Parameter

Description

SIP TOS/DiffServ Value

Time of service (ToS)/differentiated services (DiffServ) field value in UDP IP packets carrying a SIP message. Default: 0x68.

RTP ToS/DiffServ Value

Value for the ToS field of voice data packets.

Sets the priority for voice packets in data traffic.

Default: 0xb8.

SIP Settings

Parameter

Description

SIP Transport

Select the transport protocol for SIP messages:

  • UDP

  • TCP

  • TLS

  • AUTO

AUTO allows the phone to select the appropriate protocol automatically, based on the NAPTR records on the DNS server. See Configure the SIP Transport for more details.

Default: UDP

SIP Port

The phone's port number for SIP message listening and transmission.

Specify the port number here only when you are using UDP as the SIP transport protocol.

If you are using TCP, the system uses a random port within the range specified in SIP TCP Port Min and SIP TCP Port Max on the Voice > SIP tab.

If you need to specify a port of SIP proxy server, you can specify it using the Proxy field (Proxy and Registration) or the XSI Host Server field (XSI Line Service).

Default: 5060

SIP 100REL Enable

Support of 100REL SIP extension for reliable transmission of provisional responses (18x) and use of PRACK requests. Select Yes to enable.

Default: No

EXT SIP Port

The external SIP port number.

Auth Resync-Reboot

The Cisco IP Phone authenticates the sender when it receives a NOTIFY message with the following requests:

  • resync

  • reboot

  • report

  • restart

  • XML-service

Select Yes to enable.

Default: 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

The Remote-Party-ID header to use instead of the From header. Select Yes to enable.

Default: Yes

Referor Bye Delay

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

Default: 4

Refer-To Target Contact

Indicates the refer-to target. Select Yes to send the SIP Refer to the contact.

Default: No

Referee Bye Delay

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

Default: 0

Refer Target Bye Delay

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

Default: 0

Sticky 183

When 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.

Default: No

Auth INVITE

When enabled, authorization is required for initial incoming INVITE requests from the SIP proxy. To enable this feature, select Yes.

Default: No

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).

Set G729 annexb

Configure G.729 Annex B settings.

User Equal Phone

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

To: sip:+12325551234@example.com; user=phone

To enable this optional parameter, select Yes.

Default: No

Call Recording Protocol

Determines the type of recording protocol that the phone uses. Options are:

  • SIPINFO

  • SIPREC

Default: SIPREC

Privacy Header

Sets user privacy in the SIP message in the trusted network.

The privacy header options are:

  • Disabled (default)

  • none—The user requests that a privacy service applies no privacy functions to this SIP message.

  • header—The user needs a privacy service to obscure headers which cannot be purged of identifying information.

  • session—The user requests that a privacy service provide anonymity for the sessions.

  • user—The user requests a privacy level only by intermediaries.

  • id—The user requests that the system substitute an id that doesn't reveal the IP address or host name.

Default: Disabled

P-Early-Media Support

Controls whether the P-Early-Media header is included in the SIP message for an outgoing call.

To include the P-Early-Media header, select Yes. Otherwise, select No.

Default: No

Call Feature Settings

Parameter

Description

Blind Attn-Xfer Enable

Enables the 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 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.

Default: No

Message Waiting

Indicates whether the Message Waiting Indicator on the phone is lit. This parameter toggles a message from the SIP proxy to indicate if a message is waiting.

Auth Page

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

Default: No

Default Ring

Type of ring heard. Choose from No Ring or 1 through 10.

Ring options are Sunlight, Chirp 1, Chirp 2, Delight, Evolve, Mellow, Mischief, Reflections, Ringer, Ascent, Are you there, and Chime.

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.

Conference Bridge URL

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

Auth Page Password

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

Mailbox ID

Identifies the voice mailbox number/ID for the phone.

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.

Auto Ans Page On Active Call

Determines the behavior of the phone when a page call arrives.

Feature Key Sync

Enable the synchronization of settings between the line and the server if necessary.

Feature Key Sync must be enabled for lines that are configured for the following functions or users:

  • Call forward all

  • DND

Call Park Monitor Enable

BroadSoft server-only feature. If call park is enabled on the server or on any of the programmable line keys, you need to enable this field for call park notification to work.

Default: No

Enable Broadsoft Hoteling

When this parameter is set to yes, the phone sends out subscription messages (without body) to the server.

Default: No

Hoteling Subscription Expires

An expiration value that is added in the subscription message. Default value is 3600.

Secure Call Option

The secure call feature works only when the SIP Transport on the Ext (n) tab is set to TLS.

Enables secured calls on an extension. Options are:

  • Optional: The phone maintains the current behavior for secure calls.

  • Required: The phone rejects nonsecure calls from other phones.

Default: Optional

ACD Settings

Parameter

Description

Broadsoft ACD

Enables the phone for Automatic Call Distribuion (ACD). Select Yes to enable or No to disable.

Default: No

Call Information Enable

Enables the phone to display details of a call center call. Select Yes to enable or No to disable.

Default: No

Disposition Code Enable

Enables the user to add a disposition code. Select Yes to enable or No to disable.

Default: No

Trace Enable

Enables the user to trace the last incoming call. Select Yes to enable or No to disable.

Default: No

Emergency Escalation Enable

Enables the user to escalate a call to a supervisor in case of emergency. Select Yes to enable or No to disable.

Default: No

Queue Status Notification Enable

Displays the call center status and the agent status. Select Yes to enable or No to disable.

Default: No

Proxy and Registration

Parameter

Description

Proxy

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

The port number is optional. If you don't specify a port, the default port 5060 is used for UDP, and the default port 5061 is used for TLS.

When you need to refer to this proxy in another setting, for example, the speed dial line key configuration, use the $PROXY macro variable.

Outbound Proxy

All outbound requests are sent as the first hop. Enter an IP address or domain name.

Alternate Proxy

Alternate Outbound Proxy

This feature provides fast fall back 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 fall back 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 OB Proxy In Dialog

Determines whether to force SIP requests to be sent to the outbound proxy within a dialog. Ignored if the Use Outbound Proxy field is set to No or if the Outbound Proxy field is empty.

Default: Yes

Register

Enables periodic registration with the proxy. This parameter is ignored if a proxy is not specified. To enable this feature, select Yes.

Default: Yes

Make Call Without Reg

Enables making outbound calls without successful (dynamic) registration by the phone. If set to No, the dial tone plays only when registration is successful. To enable this feature, select Yes.

Default: No

Register Expires

Defines how often the phone renews registration with the proxy. If the proxy responds to a REGISTER with a lower expires value, the phone renews registration based on that lower value instead of the configured value.

If registration fails with an “Expires too brief” error response, the phone retries with the value specified in the Min-Expires header of the error.

The range is from 32 to 2000000.

Default: 3600 seconds

Ans Call Without Reg

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

Default: No

Use DNS SRV

Enables DNS SRV lookup for the proxy and outbound proxy. To enable this feature, select Yes. Otherwise, select No.

Default: No

DNS SRV Auto Prefix

Enables the phone to automatically prepend the proxy or outbound proxy name with _sip._udp when performing a DNS SRV lookup on that name.

Default: No

Proxy Fallback Intvl

Sets the delay after which the phone retries from the highest priority proxy (or outbound proxy) after it has failed over to a lower priority server.

The phone should have the primary and backup proxy server list from a DNS SRV record lookup on the server name. It needs to know the proxy priority; otherwise, it does not retry.

The range is from 0 to 65535.

Default: 3600 seconds

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 Port, the phone uses normal, then inspects the port number based on the first-listed proxy port.

Default: Normal

Dual Registration

Set to Yes to enable the Dual registration/Fast Fall back feature. To enable the feature you must also configure the alternate proxy/alternate outbound proxy fields in the Proxy and Registration section.

Auto Register When Failover

If set to No, the fallback happens immediately and automatically. If the Proxy Fallback Intvl is exceeded, all the new SIP messages go to the primary proxy.

If set to Yes, the fallback happens only when current registration expires, which means only a REGISTER message can trigger fallback.

For example, when the value for Register Expires is 3600 seconds and Proxy Fallback Intvl is 600 seconds, the fallback is triggered 3600 seconds later and not 600 seconds later. When the value for Register Expires is 600 seconds and Proxy Fallback Intvl is 1000 seconds, the fallback is triggered at 1200 seconds. After successfully registering back to primary server, all the SIP messages go to primary server.

Subscriber Information

Parameter

Description

Display Name

Name displayed as the caller ID.

User ID

Extension number for this line.

When you need to refer to this user ID in another setting, for example, the short name for a line key, use the $USER macro variable.

Password

Password for this line.

Default: Blank (no password required)

Auth ID

Authentication ID for SIP authentication.

Default: Blank

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 if a SIP or SIPS URI contains no @ character, 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.

XSI Line Service

Parameter

Description

XSI Host Server

Enter the name of the server; for example, .

xsi.iop1.broadworks.net

Note 

XSI Host Server uses http protocol by default. To enable XSI over HTTPS, you can specify https:// in the server.

For example:

https://xsi.iop1.broadworks.net

You can also specify a port for the server.

For example:

https://xsi.iop1.broadworks.net:5061

If you don't specify a port. The default port for the specified protocol is used.

Default: Blank

XSI Authentication Type

Determines the XSI authentication type. Select Login Credentials to authenticate access with Login User ID and Login Password. Select SIP Credentials to authenticate access with the register Auth ID and Password of the SIP account registered on the phone.

Default: Login Credentials

Login User ID

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

For any XSI Authentication Type, you must enter Login User ID. Without Login User ID, the BroadWorks Anywhere feature does not work.

Default: Blank

Login Password

Alphanumeric password associated with the Login User ID.

Enter Login Password, when you select Login Credentials for XSI authentication type.

Default: Blank

Anywhere Enable

Enables BroadWorks Anywhere feature on an extension.

If you choose Yes, Anywhere is enabled on this line, and the user can use the phone menu to add multiple locations to this specific line.

Default: Yes

Block CID Enable

Enables XSI Caller ID blocking on a line.

Choose Yes to enable the synchronization of blocking caller id status with the server using XSI interface. Choose No to use the phone's local blocking caller id settings.

CFWD Enable

Enables or disables call forwarding status sync on a line via XSI service.

Choose Yes to enable the phone to synchronize the call forwarding status with the server using the XSI service. Choose No to disable this feature.

Note 
  • When Feature Key Sync is set to Yes, FKS takes precedent over XSI synchronization.

  • If XSI host server and credentials are not entered and the CFWD Enable field is set to Yes, the phone user can't forward calls on the phone.

DND Enable

Enables or disables DND status sync on a line via XSI service.

Choose Yes to enable the phone to synchronize DND status with the server using the XSI service. Choose No to disable this feature.

Note 
  • When Feature Key Sync is set to Yes, FKS takes precedent over XSI synchronization.

  • If XSI host server and credentials are not entered and the DND Enable field is set to Yes, the phone user can't turn on DND mode on the phone.

Audio Configuration

Parameter

Description

Preferred Codec

Preferred codec for all calls. The actual codec used in a call still depends on the outcome of the codec negotiation protocol.

Select one of the following:

  • G711u

  • G711a

  • G729a

  • G729ab

  • G722

  • G722.2

  • iLBC

  • OPUS

  • iSAC

Default: G711u

Use Pref Codec Only

Select No to use any code. Select Yes to use only the preferred codes. When you select Yes, calls fail if the far end does not support the preferred codecs.

Default: No

Second Preferred Codec

Codec to use if the first codec fails.

Default: Unspecified

Third Preferred Codec

Codec to use if the second codec fails.

Default: Unspecified

G711u Enable

Enables use of the G.711u codec.

Default: Yes

G711a Enable

Enables use of the G.711a codec.

Default: Yes

G729a Enable

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

Default: Yes

G722 Enable

Enables use of the G.722 codec.

Default: Yes

G722.2 Enable

Enables use of the G.722.2 codec.

Default: No

iLBC Enable

Enables use of the iLBC codec.

Default: Yes

OPUS Enable

Enables the use of OPUS codec.

Default: Yes

Silence Supp Enable

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

Default: No

DTMF Tx Method

The method for transmitting DTMF signals to the far end. The options are:

  • AVT—Audio video transport. Sends DTMF as AVT events.

  • InBand—Sends DTMF by using the audio path.

  • Auto—Uses InBand or AVT based on the outcome of codec negotiation.

  • INFO—Uses the SIP INFO method.

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.

Encryption Method

Encryption method to be used during secured call. Options are AES 128 and AES 256 GCM

Default: 128.

Dial Plan

Parameter

Description

Dial Plan

Dial plan script for the selected extension.

The dial plan syntax allows the designation of three parameters for use with a specific gateway:
  • uid – The authentication user-id

  • pwd – The authentication password

  • nat – If this parameter is present, use NAT mapping.

Separate each parameter with 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.

Enable URI Dialing

Enables or disables URI dialing.

Emergency Number

Enter a comma-separated list of emergency numbers. When one of these numbers is dialed, the unit disables processing of CONF, HOLD, and other similar softkeys or buttons to avoid accidentally putting the current call on hold. The phone also disables hook flash event handling.

Only the far end can terminate an emergency call. The phone is restored to normalcy after the call is terminated and the receiver is back on-hook.

Maximum number length is 63 characters. Defaults to blank (no emergency number).

E911 Geolocation Configuration

E911 Geolocation Configuration

Parameter

Description

Company UUID

The Universally Unique Identifier (UUID) assigned to the customer by the emergency call services provider.

Maximum identifier length is 128 characters. Defaults to blank.

Primary Request URL

Encrypted HTTPS phone location request. The request uses the phone IP addresses, MAC address, Network Access Identifier (NAI), and chassis ID and port ID assigned by the network switch manufacturer. The request also includes the location server name and the customer identifier.

The server used by the emergency call services provider responds with an Emergency Response Location (ERL) that has a location Uniform Resource Identifier (URI) tied to the user phone IP address.

Defaults to blank.

Secondary Request URL

Encrypted HTTPS request sent to the emergency call services provider's backup server to obtain the user's phone location.

Defaults to blank.

See Emergency Call Support Terminology for terms that describe emergency call support for phones.

User

Hold Reminder

Parameter

Description

Hold Reminder Timer

Specifies the time delay (in seconds), that a ring splash is heard on an active call when another call was placed on hold.

Default: 0

Hold Reminder Ringtone

Specifies the volume of the timer ringtone.

Call Forward

Parameter

Description

Cfwd Setting

Select Yes to enable call forwarding.

Cfwd All Dest

Enter the extensions to which the call is forwarded.

Cfwd Busy Dest

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

Default: voicemail

Cfwd No Ans Dest

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

Default: voicemail

Cfwd No Ans Delay

Enter the delay in time (in seconds) to wait before forwarding a call that is unanswered.

Default: 20 seconds

Speed Dial

Parameter

Description

Speed Dial Name (2 to 9)

Name assigned to a specific speed dial number.

Default: Blank

Speed Dial Number 2 to 9)

Target phone number (or URL) assigned to speed dial 2, 3, 4, 5, 6, 7, 8, or 9. Press the digit key (2-9) to dial out the assigned number.

Default: Blank

Supplementary Services

Parameter

Description

CW Setting

Enables or disables the Call Waiting service.

Default: Yes

Block CID Setting

Enables or disables the Block CID service.

Default: No

Block ANC Setting

Enables or disables the Block ANC service.

Default: No

DND Setting

Enables or disables the DND settings options for a user.

Handset LED Alert

Enables or disables LED alert on the handset. Options are: Voicemail and Voicemail, Missed Call.

Default: Voicemail

Secure Call Setting

Enables or disables Secure Call.

Default: No

Auto Answer Page

Enables or disables automatic answering of paged calls.

Default: Yes

Preferred Audio Device

Choose the type of audio that the phone will use. Options are: Speaker and Headset.

Choose the type of audio that the phone will use. Options are: Speaker and Headset.

Default: None

Time Format

Choose the time format for the phone (12 or 24 hour).

Default: 12hr

Date Format

Choose the date format for the phone (month/day or day/month).

Default: month/day

Miss Call Shortcut

Enables or disables the option for creating a missed call shortcut.

Alert Tone Off

Enables or disables the alert tone.

Log Missed Calls for EXT (n)

Enables or disables the missed calls logs for a specific extension.

Shared Line DND Cfwd Enable

Enable/disable the Shared Line DND Call Forward.

Audio Volume

Parameter

Description

Ringer Volume

Sets the default volume for the ringer.

Default: 9

Speaker Volume

Sets the default volume for the speakerphone.

Default: 8

Handset Volume

Sets the default volume for the handset.

Default: 10

Headset Volume

Sets the default volume for the headset.

Default: 10

Electronic Hookswitch Control

Enables or disables the Electronic HookSwitch (EHS) feature.

After EHS is enabled, the AUX port does not output phone logs.

Audio Compliance

Parameter

Description

Compliant Standard

Specifies the compliance standard for the phone audio. The available options are:

  • ETSI: A set of standards for speech and multimedia transmission for the narrowband and wideband terminals from European Telecommunications Standards Institute (ETSI).

  • TIA: A set of standards from US Telecommunications Industry Association (TIA). The standards are for narrowband and wideband audio transmission via wired telephones.

Default: TIA

Screen

Parameter

Description

Screen Saver Enable

Enables a screen saver on the phone. When the phone is idle for a specified time, it enters screen saver mode.

Default: No

Screen Saver Type

Types of screen saver. Options you can choose:
  • Clock: Displays a digital clock on a plain background.

  • Download Picture: Displays a picture pushed from the phone webpage.

  • Lock : Enables locking of the screensaver.

Screen Saver Wait

Amount of idle time before screen saver displays.

enter the number of seconds of idle time to elapse before the screen saver starts.

Default: 300

Screen Saver Refresh Period

Number of seconds before the screen saver should refresh (if, for example, you chose a rotation of pictures).

Back Light Timer

Number of seconds for which the back light timer will be on.

LCD Contrast

Value for desired contrast.

Logo Type

Type of logo displayed on the phone screen. Options you can choose:

  • Default

  • Download Picutre

  • Text Logo

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:
Super
Telecom

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

Picture Download URL

URL locating the (.png) file to display on the phone screen background.

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

Att Console

General


Note

The attendant console tab, labeled Att Console, is only available in Admin Login > advanced mode.


Parameter

Description

Subscribe Expires

Specifies how long the subscription remains valid. After the specified period of time elapses, the Cisco Attendant Console initiates a new subscription.

Default: 1800

Subscribe Retry Interval

Specifies the length of time to wait to try again if the subscription fails.

Default: 30

Subscribe Delay

Length of delay before attempting to subscribe.

Default: 1

Server Type

Specifies the server type with which the phone is connected.

Options available:

  • BroadSoft

  • SPA9000

  • Asterisk

  • RFC3265_4235

  • Sylantro

BLF List URI

The Uniform Resource Identifier (URI) of the Busy Lamp Field (BLF) list that you have set up for a user of the phone, on the BroadSoft server.

This field is only applicable if the phone is registered to a BroadSoft server. The BLF list is the list of users whose lines the phone is allowed to monitor. See Phone Configuration for Monitoring Other Phones for details.

The BLF List URI must be specified in the format <URI name>@<server>. The BLF List URI specified must be the same as the value configured for the List URI: sip parameter on the BroadSoft server.

Default: Blank

Example XML configuration:

<BLF_List_URI ua="na">MonitoredUsersList@sipurash22.com</BLF_List_URI>
Use Line Keys For BLF List

Controls whether the phone uses its line keys to monitor the BLF list, when monitoring of the BLF list is active.

This setting only has significance when BLF List is set to Show.

Default: No

Example XML configuration:

<Use_Line_Keys_For_BLF_List ua="na">Yes</Use_Line_Keys_For_BLF_List>
Customizable PLK Options

Features that users are allowed to configure on line keys.

To allow a feature, add the corresponding option as shown below. Separate options with the semi-colon (;).

  • Speed dial: sd

  • Busy Lamp Field (BLF) key to monitor a user: blf

  • Call pickup from a monitored line: cp

    Note 
    This option is only effective when the blf option is added.

Default: sd;

Note 

Adding the sd option automatically allows users to configure speed dial to a monitored line (speed dial with BLF) when the blf option is added.

Example, to allow all features:

sd;blf;cp

Example XML configuration:

<Customizable_PLK_options ua="na">sd;</Customizable_PLK_options>
BLF List

Activates or deactivates monitoring of the BLF list.

When set to Show, the phone assigns available line keys in sequence, to monitor the BLF list entries. The labels of the BLF list keys show the names of the monitored users and the status of the monitored lines.

This setting only has significance when BLF List URI is configured.

Example XML configuration:

<BLF_List ua="rw">Show</BLF_List>

BXfer to Starcode Enable

When set to Yes, the phone performs a blind transfer when the *code is defined in a speed dial extended function,. If set to No, the current call is held and a new call is started to the speed dial destination.

Default: No

BXfer On Speed Dial Enable

When set to Yes, the phone performs a blind transfer when the speed dial function key is selected. When set to no, the current connected call is held and a new call to the speed dial destination is started.

For example, when a user parks a call using the speed dial function, if the parameter is enabled, a blind transfer is performed to the parking lot. If the parameter is not enabled, an attended transfer is performed to the parking lot.

Default: No

BXfer To Remote Party Number Enable

When set to Yes, the phone performs a blind transfer to a remote number. When set to no, blind transfer to remote number is disabled.

BLF Label Display Mode

Options to select a mode which displays on the phone screen for BLF.

Default: Blank

Unit

Enter the programming information for each line key for the Attendant Console unit.

Parameter

Description

Unit Enable

Indicates whether the key expansion module that is added to the phone is enabled.

Unit Online

Indicates whether the key expansion module that is added to the phone is active.

HW Version

Displays the hardware version of the key expansion module that is added to the phone..

SW Version

Displays the software version of the key expansion module that is added to the phone.

TR-069

TR-069

Parameter

Description

Enable TR-069

Settings that enables or disables the TR-069 function.

Default: Disabled

ACS URL

URL of the ACS that uses the CPE WAN Management Protocol. This parameter must be in the form of a valid HTTP or HTTPS URL. The host portion of this URL is used by the CPE to validate the ACS certificate when it uses SSL or TLS.

ACS Username

Username that authenticates the CPE to the ACS when ACS uses the CPE WAN Management Protocol. This username is used only for HTTP-based authentication of the CPE.

If the user name is not configured, admin is used as default.

ACS Password

Password to access to the ACS for a specific user. This password is used only for HTTP-based authentication of the CPE.

If the password is not configured, admin is used as default.

ACS URL In Use

URL of the ACS that is currently in use. This is a read-only field.

Connection Request URL

URL of the ACS that makes the connection request to the CPE.

Connection Request Username

Username that authenticates the ACS that makes the connection request to the CPE.

Connection Request Password

Password used to authenticate the ACS that makes a connection request to the CPE.

Periodic Inform Interval

Duration in seconds of the interval between CPE attempts to connect to the ACS when Periodic Inform Enable is set to yes.

Default value is 20 seconds.

Periodic Inform Enable

Settings that enables or disables the CPE connection requests. Default value is Yes.

TR-069 Traceability

Settings that enables or disables TR-069 transaction logs.

The default value is No.

CWMP V1.2 Support

Settings that enables or disables CPE WAN Management Protocol (CWMP) support. If set to disable, the phone does not send any Inform messages to the ACS nor accept any connection requests from the ACS.

Default value is Yes.

TR-069 VoiceObject Init

Settings to modify voice objects. Select Yes to initialize all voice objects to factory default values or select No to retain the current values.

TR-069 DHCPOption Init

Settings to modify DHCP settings. Select Yes to initialize the DHCP settings from the ACS or select No to retain the current DHCP settings.

TR-069 Fallback Support

Settings that enables or disables the TR-069 fallback support.

If the phone attempts to discover the ACS with DHCP and is unsuccessful, the phone next uses DNS to resolve the ACS IP address.

BACKUP ACS URL

Backup URL of the ACS that uses the CPE WAN Management Protocol. This parameter must be in the form of a valid HTTP or HTTPS URL. The host portion of this URL is used by the CPE to validate the ACS certificate when it uses SSL or TLS.

BACKUP ACS User

Backup username that authenticates the CPE to the ACS when ACS uses the CPE WAN Management Protocol. This username is used only for HTTP-based authentication of the CPE.

BACKUP ACS Password

Backup password to access to the ACS for a specific user. This password is used only for HTTP-based authentication of the CPE.

Note 

If you do not configure the above parameters, you can also fetch them through DHCP options 60,43, and 125.

Call History

Displays the call history for the phone. To change the information displayed, select the type of call history from the following tabs:

  • All Calls

  • Missed

  • Received

  • Placed

Select Add to Directory to add the call information to your Personal Directory.

Personal Directory

The Personal Directory allows a user to store a set of personal numbers. Directory entries can include the following contact information:

  • No. (the directory number)

  • Name

  • Work

  • Mobile

  • Home

  • Speed Dials

To edit contact information, click Edit Contacts.