Cisco Small Business SPA100 Series Phone Adapters Administration Guide

Product Name

Model number/name.

Serial Number

Product serial number.

Software Version

Software version number.

Hardware Version

Hardware version number.

MAC Address

MAC Address. For example: 8843E1657936.

Client Certificate

Status of the client certificate, which can indicate if the ATA was authorized by your ITSP.

Customization

Used for Remote Configuration by service providers who deploy the ATA to their customers.

• Open: Not a Remote Configuration unit. This ATA can be configured by using the configuration utility.
• Pending: This Remote Configuration unit has not yet connected to the server to get its profile.
• Customized Value: This Remote Configuration unit has received its profile from the server. For Example: ISP_Name

Current Time

Current date and time of the system; for example, 10/3/2003 16:43:00. Set the system time by using the Network Setup > Time Settings page.

Elapsed Time

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

RTP Packets Sent

Total number of RTP packets sent (including redundant packets)

RTP Bytes Sent

Total number of RTP bytes sent.

RTP Packets Recv

Total number of RTP packets received (including redundant packets)

RTP Bytes Recv

Total number of RTP bytes received.

SIP Messages Sent

Total number of SIP messages sent (including retransmissions)

SIP Bytes Sent

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

SIP Messages Recv

Total number of SIP messages received (including retransmissions)

SIP Bytes Recv

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

External IP

The External IP address used for NAT mapping.

Hook State

The hook state of the port: On or Off.

Registration State

Indicates if the line has registered with the SIP proxy.

Last Registration At

Last date and time the line was registered.

Next Registration In

The number of seconds before the next registration renewal.

Message Waiting

Indicates whether you have new voice mail waiting.

Indicates Yes when a message is received.

Mapped SIP Port

Port number of the SIP port mapped by NAT.

Call Back Active

Indicates whether or not a call back request is in progress. Options are either yes or no.

Last Called Number

The phone number that was most recently called through this port.

Last Caller Number

The originating phone number of the call that was most recently received through this port.

Call 1 and 2 State

Indicates the state of calls.

Call 1 and 2 Tone

The type of tone used by the call.

Call 1 and 2 Encoder

The codec used for encoding.

Call 1 and 2 Decoder

The codec used for decoding.

Call 1 and 2 FAX

The status of the fax mode.

Call 1 and 2 Type

The direction of the call. May take one of the following values:

• Inbound
• Outbound

Call 1 and 2 Remote Hold

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

Call 1 and 2 Callback

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

Call 1 and 2 Peer Name

The name of the peer phone.

Call 1 and 2 Peer Phone

The phone number of the peer phone.

Call 1 and 2 Duration

The duration of the call.

Call 1 and 2 Packets Sent

The number of packets sent

Call 1 and 2 Packets Recv

The number of packets received.

Call 1 and 2 Bytes Sent

The number of bytes sent.

Call 1 and 2 Bytes Recv

The number of bytes received.

Call 1 and 2 Decode Latency

The number of milliseconds for decoder latency.

Call 1 and 2 Jitter

The number of milliseconds for receiver jitter

Call 1 and 2 Round Trip Delay

The number of milliseconds for delay.

Call 1 and 2 Packets Lost

The number of packets lost.

Call 1 and 2 Packet Error

The number of invalid packets received.

Custom CA Provisioning Status

The status of the latest custom CA (Certificate Authority) certificate download.

Custom CA Info

The successfully downloaded CA information, or “Not Installed” if no custom CA certificate was installed.
Default setting: Not Installed

Restricted Access Domains

Domain that Cisco IP phones responds to SIP messages only from the identified servers. Applicable to Line 1.

IVR Admin Passwd

Password for the administrator to manage the ATA by using the built-in IVR through a connected phone.

Network Startup Delay

The number of seconds of delay between restarting the voice module and initializing network interface.
Default setting: 3

DNS Query TTL Ignore

In DNS packages, the server will suggest a TTL value to the client; if this parameter is set to yes, the value from the server will be ignored.
Default setting: no

Syslog Server

Specify the syslog server domain name/IP and port. This feature specifies the server for logging ATA system information and critical events. If both Debug Server and Syslog Server are specified, Syslog messages are also logged to the Debug Server.
Default setting: blank

Debug Server

The debug server domain name/IP and port. This feature specifies the server for logging debug information. The level of detailed output depends on the debug level parameter setting.
Default setting: blank

Debug Level

Determines the level of debug information that will be generated. Select 0, 1, 2, 3, 3+Router, 3+Coma or 1+Coma from the drop-down list. The higher the debug level, the more debug information will be generated. Level 0 means that no information will be collected. Levels 1, 2 & 3 generate messages related to the voice ports only. Level 3+Router generates debug content for both voice and router components.
Levels 1/3+ Coma generates vendor’s cordless debug content and voice component.
Default setting: 3

Syslog Prefix

Prefixes additional information to syslog with configuration set.
Default setting: blank

Max Forward

The maximum times a call can be forwarded. The valid range is from 1 to 255.

Default setting: 70

Max Redirection

Number of times an invite can be redirected to avoid an infinite loop.
Default setting: 5.

Max Auth

The maximum number of times (from 0 to 255) a request may be challenged.
Default setting: 2

SIP User Agent Name

The User-Agent header used in outbound requests. If empty, the header is not included. Macro expansion of $A to $D corresponding to GPP_A to GPP_D allowed.
Default setting: $VERSION

SIP Server Name

The server header used in responses to inbound responses.
Default setting: $VERSION

SIP Reg User Agent Name

The User-Agent name to be used in a REGISTER request. If this value is not specified, the SIP User Agent Name parameter is also used for the REGISTER request.
Default setting: blank

SIP Reg Starting Sequence Number

Defines the SIP Reg message Sequence Number.
Default setting: blank

SIP Accept Language

Accept-Language header used. There is no default (this indicates that the ATA does not include this header) If empty, the header is not included.
Default setting: blank

DTMF Relay MIME Type

The MIME Type used in a SIP INFO message to signal a DTMF event.
Default setting: application/dtmf-relay.

Hook Flash MIME Type

The MIME Type used in a SIP INFO message to signal a hook flash event.
Default setting: application/hook-flash

Remove Last Reg

Determines whether or not the ATA removes the last registration before submitting a new one, if the value is different. Select yes to remove the last registration, or select no to omit this step.
Default setting: no

Use Compact Header

Determines whether or not the ATA uses compact SIP headers in outbound SIP messages. Select yes or no from the drop-down list. Select yes to use compact SIP headers in outbound SIP messages. Select no to use normal SIP headers. If inbound SIP requests contain compact headers, the ATA reuses the same compact headers when generating the response regardless the settings of the Use Compact Header parameter. If inbound SIP requests contain normal headers, the ATA substitutes those headers with compact headers (if defined by RFC 261) if Use Compact Header parameter is set to yes.
Default setting: no

Escape Display Name

Determines whether or not the Display Name is private. Select yes if you want the ATA to enclose the string (configured in the Display Name) in a pair of double quotes for outbound SIP messages. If the display name includes " or \, these will be escaped to \" and \\ within the double quotes. Otherwise, select no.
Default setting: no

RFC 2543 Call Hold

Configures the type of call hold: a:sendonly or 0.0.0.0. Do not use the 0.0.0.0 syntax in a HOLD SDP; use the a:sendonly syntax.
Default setting: yes

Mark all AVT Packets

Select yes if you want all AVT tone packets (encoded for redundancy) to have the marker bit set for each DTMF event. Select no to have the marker bit set only for the first packet.
Default setting: yes

AVT Packet Size

Indicates the AVT Packet size according to value set in ptime or fixed 10ms.
Default setting: ptime

SIP TCP Port Min

The lowest TCP port number that can be used for SIP sessions.
Default setting: 5060

SIP TCP Port Max

The highest TCP port number that can be used for SIP sessions.
Default setting: 5080

CTI Enable

Enables or disables the Computer Telephone Interface feature provided by some servers.
Default setting: no

Keep Referee When REFER Failed

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

You can also configure this parameter in the configuration file:

<Keep_Referee_When_REFER_Failed ua="na">Yes

</Keep_Referee_When_REFER_Failed>

Caller ID Header

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

SIP T1

RFC 3261 T1 value (round-trip time estimate), which can range from 0 to 64 seconds.
Default setting: 0.5

SIP T2

RFC 3261 T2 value (maximum retransmit interval for non-INVITE requests and INVITE responses), which can range from 0 to 64 seconds.
Default setting: 4

SIP T4

RFC 3261 T4 value (maximum duration a message remains in the network), which can range from 0 to 64 seconds.
Default setting: 5

SIP Timer B

INVITE time-out value, which can range from 0 to 64 seconds.
Default setting: 32

SIP Timer F

Non-INVITE time-out value, which can range from 0 to 64 seconds.
Default setting: 16

SIP Timer H

H INVITE final response, time-out value, which can range from 0 to 64 seconds.
Default setting: 32

SIP Timer D

ACK hang-around time, which can range from 0 to 64 seconds.
Default setting: 32

SIP Timer J

Non-INVITE response hang-around time, which can range from 0 to 64 seconds.
Default setting: 32

INVITE Expires

INVITE request Expires header value. If you enter 0, the Expires header is not included in the request. Range: 0–(231–1)
Default setting: 240

ReINVITE Expires

ReINVITE request Expires header value. If you enter 0, the Expires header is not included in the request. Range: 0–(231–1)
Default setting: 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.
Default setting: 1

Reg Max Expires

Maximum registration expiration time allowed from the proxy in the Min-Expires header. If the value is larger than this setting, the maximum value is used.
Default setting: 7200

Reg Retry Intvl

Interval to wait before the ATA retries registration after failing during the last registration.
Default setting: 30

Reg Retry Long Intvl

When registration fails with a SIP response code that does not match Retry Reg RSC, the ATA waits for the specified length of time before retrying. If this interval is 0, the ATA stops trying. This value should be much larger than the Reg Retry Intvl value, which should not be 0.
Default setting: 1200

Reg Retry Random Delay

Random delay range (in seconds) to add to Register Retry Intvl when retrying REGISTER after a failure.
Default setting: 0 (disabled)

Reg Retry Long Random Delay

Random delay range (in seconds) to add to Register Retry Long Intvl when retrying REGISTER after a failure.
Default setting: 0 (disabled)

Reg Retry Intvl Cap

The maximum value to cap the exponential back-off retry delay (which starts at Register Retry Intvl and doubles on every REGISTER retry after a failure) In other words, the retry interval is always at Register Retry Intvl seconds after a failure. If this feature is enabled, Reg Retry Random Delay is added on top of the exponential back-off adjusted delay value.
Default setting: 0, which disables the exponential backoff feature.

SIT1 RSC

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

SIT2 RSC

SIP response status code to INVITE on which to play the SIT2 Tone.
Default setting: blank

SIT3 RSC

SIP response status code to INVITE on which to play the SIT3 Tone.
Default setting: blank

SIT4 RSC

SIP response status code to INVITE on which to play the SIT4 Tone.
Default setting: blank

Try Backup RSC

SIP response code that retries a backup server for the current request.
Default setting: blank

Retry Reg RSC

Interval to wait before the ATA retries registration after failing during the last registration.
Default setting: blank

RTP Port Min

Minimum port number for RTP transmission and reception.

The RTP Port Min and RTP Port Max parameters should define a range that contains at least 4 even number ports, such as 100 –106.
Default setting: 16384.

RTP Port Max

Maximum port number for RTP transmission and reception.
Default setting: 16482.

RTP Packet Size

Packet size in seconds, which can range from 0.01 to 0.16. Valid values must be a multiple of 0.01 seconds.
Default setting: 0.030

RTP Tx Packet Size Follows Remote SDP

Enable the Remote pair RTP Packet Size.
Default setting: Yes

Max RTP ICMP Err

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

RTCP Tx Interval

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

No UDP Checksum

Select yes if you want the ATA to calculate the UDP header checksum for SIP messages. Otherwise, select no.
Default setting: no

Stats In BYE

Determines whether the ATA includes the P-RTP-Stat header or response in a BYE message. The header contains the RTP statistics of the current call. Select yes or no from the drop-down list.
Default setting: yes

The format of the P-RTP-Stat header is:

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

NSE Dynamic Payload

NSE dynamic payload type. The valid range is 96-127.
Default setting: 100

AVT Dynamic Payload

AVT dynamic payload type. The valid range is 96-127.
Default setting: 101

INFOREQ Dynamic Payload

INFOREQ dynamic payload type.
Default setting: blank

G726r32 Dynamic Payload

G726r32 dynamic payload type.
Default setting: 2

G729b Dynamic Payload

G.729b dynamic payload type. The valid range is 96-127.
Default setting: 99

EncapRTP Dynamic Payload

EncapRTP Dynamic Payload type.
Default setting: 112

RTP-Start-Loopback Dynamic Payload

RTP-Start-Loopback Dynamic Payload type.
Default setting: 113

RTP-Start-Loopback Codec

RTP-Start-Loopback Codec. Select one of the following: G711u, G711a, G726-32, G729a.
Default setting: G711u

NSE Codec Name

NSE codec name used in SDP.
Default setting: NSE

AVT Codec Name

AVT codec name used in SDP.
Default setting: telephone-event

G711u Codec Name

G.711u codec name used in SDP.
Default setting: PCMU

G711a Codec Name

G.711a codec name used in SDP.
Default setting: PCMA

G726r32 Codec Name

G.726-32 codec name used in SDP.
Default setting: G726-32

G729a Codec Name

G.729a codec name used in SDP.
Default setting: G729a

G729b Codec Name

G.729b codec name used in SDP.
Default setting: G729ab

EncapRTP Codec Name

EncapRTP codec name used in SDP.
Default setting: encaprtp

Handle VIA received

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

Handle VIA rport

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

Insert VIA received

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

Insert VIA rport

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

Substitute VIA Addr

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

Send Resp To Src Port

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

STUN Enable

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

STUN Test Enable

If the STUN Enable feature is enabled and a valid STUN server is available, the ATA can perform a NAT-type discovery operation when it powers on. It contacts the configured STUN server, and the result of the discovery is reported in a Warning header in all subsequent REGISTER requests. If the ATA detects symmetric NAT or a symmetric firewall, NAT mapping is disabled.
Default setting: no

STUN Server

IP address or fully-qualified domain name of the STUN server to contact for NAT mapping discovery.
Default setting: blank

EXT IP

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

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

This option requires that you have (1) a static IP address from your Internet Service Provider and (2) an edge device with a symmetric NAT mechanism. If the ATA is the edge device, the second requirement is met.
Default setting: blank

EXT RTP Port Min

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

NAT Keep Alive Intvl

Interval between NAT-mapping keep alive messages.
Default setting: 15

Redirect Keep Alive

Enables or disables NAT Redirect keep alive messages.
Default setting: no

Linksys Key System

To enable operation with the Cisco SPA9000, choose yes. Otherwise, choose no.
Default setting: no

Multicast Address

The multicast address for devices in the Cisco SPA9000 voice network.
Default setting: 224.168.168.168:6061

Key System Auto Discovery

To enable auto-discovery of the Cisco SPA9000 voice system, choose yes. Otherwise, choose no.
Default setting: yes

Key System IP Address

The IP address of the Cisco SPA9000.
Default setting: blank

Force LAN Codec

If needed, specify a voice codec.
Default setting: none

Station Name

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

Provision Enable

Controls all resync actions independently of firmware upgrade actions. Set to yes to enable remote provisioning.
Default setting: yes

Resync On Reset

Triggers a resync after every reboot except for reboots caused by parameter updates and firmware upgrades.
Default setting: yes

Resync Random Delay

The maximum value for a random time interval that the ATA waits before making its initial contact with the provisioning server. This delay is effective only on the initial configuration attempt following power-on or reset. The delay is a pseudo-random number between zero and this value.

This parameter is in units of 20 seconds; the default value of 2 represents 40 seconds. This feature is disabled when this parameter is set to zero.

This feature can be used to prevent an overload of the provisioning server when a large number of devices power-on simultaneously.
Default setting: 2 (40 seconds)

Resync At (HHmm)

The time of day when the device tries to resync. The resync is performed each day. Used in conjunction with the Resync At Random Delay.
Default setting: blank

Resync At Random Delay

Used in conjunction with the Resync At (HHmm) setting, this parameter sets a range of possible values for the resync delay. The system randomly chooses a value from this range and waits the specified number of seconds before attempting to resync. This feature is intended to prevent the network jam that would occur if all resynchronizing devices began the resync at the exact same time of day.
Default setting: 600

Resync Periodic

The time interval between periodic resyncs with the provisioning server. The associated resync timer is active only after the first successful synchronization with the server. Setting this parameter to zero disables periodic resynchronization.
Default setting: 3600

Resync Error Retry Delay

Resync retry interval (in seconds) applied in case of resync failure.

The ATA has an error retry timer that activates if the previous attempt to sync with the provisioning server fails. The ATA waits to contact the server again until the timer counts down to zero.

This parameter is the value that is initially loaded into the error retry timer. If this parameter is set to zero, the ATA immediately retries to sync with the provisioning server following a failed attempt.
Default setting: 3600

Forced Resync Delay

Maximum delay (in seconds) that the ATA waits before performing a resync.

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

The ATA 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.
Default setting: 14400

Resync From SIP

Enables a resync to be triggered via a SIP NOTIFY message.
Default setting: yes

Resync After Upgrade Attempt

Triggers a resync after every firmware upgrade attempt.
Default setting: yes

Resync Trigger 1

Resync Trigger 2

Configurable resync trigger conditions. A resync is triggered when the logic equation in these parameters evaluates to TRUE.
Default setting: blank

Resync Fails On FNF

Determines whether a file-not-found response from the provisioning server constitutes a successful or a failed resync. A failed resync activates the error resync timer.
Default setting: yes

Profile Rule

This parameter is a profile script that evaluates to the provisioning resync command. The command is a TCP/IP operation and an associated URL. The TCP/IP operation can be TFTP, HTTP, or HTTPS.

If the command is not specified, TFTP is assumed, and the address of the TFTP server is obtained through DHCP option 66. In the URL, either the IP address or the FQDN of the server can be specified. The file name can have macros, such as $MA, which expands to the ATA MAC address.
Default setting: /spa$PSN.cfg

Profile Rule B:

Profile Rule C:

Profile Rule D

Defines second, third, and fourth resync commands and associated profile URLs. These profile scripts are executed sequentially after the primary Profile Rule resync operation has completed. If a resync is triggered and Profile Rule is blank, Profile Rule B, C, and D are still evaluated and executed.
Default setting: blank

DHCP Option To Use

DHCP Options, delimited by commas, retrieves firmware and profiles.
Default setting: 66.160.159.150

Transport Protocol

Transport Protocol retrieves firmware and profiles. If none is selected, TFTP is assumed and the IP address of the TFTP server is obtained from the DHCP server.
Default setting: https

Log Resync Request Msg

This parameter contains the message that is sent to the Syslog server at the start of a resync attempt.
Default setting: $PN $MAC -- Requesting resync $SCHEME://$SERVIP:$PORT$PATH

Log Resync Success Msg

Syslog message issued upon successful completion of a resync attempt.
Default setting: $PN $MAC -- Successful resync $SCHEME://$SERVIP:$PORT$PATH

Log Resync Failure Msg

Syslog message issued after a failed resync attempt.
Default setting: $PN $MAC -- Resync failed: $ERR

Report Rule

The target URL to which configuration reports are sent. This parameter has the same syntax as the Profile_Rule parameter, and resolves to a TCP/IP command with an associated URL.

A configuration report is generated in response to an authenticated SIP NOTIFY message, with Event: report. The report is an XML file containing the name and value of all the device parameters.

This parameter may optionally contain an encryption key. For example:

[ --key $K ] tftp://ps.callhome.net/$MA/rep.xml.enc
Default setting: blank

Upgrade Enable

Determines whether or not firmware upgrade operations can occur independently of resync actions.
Default setting: yes

Upgrade Error Retry Delay

The upgrade retry interval (in seconds) applied in case of upgrade failure. The ATA has a firmware upgrade error timer that activates after a failed firmware upgrade attempt. The timer is initialized with the value in this parameter. The next firmware upgrade attempt occurs when this timer counts down to zero.
Default setting: 3600

Downgrade Rev Limit

Enforces a lower limit on the acceptable version number during a firmware upgrade or downgrade. The ATA does not complete a firmware upgrade operation unless the firmware version is greater than or equal to this parameter.
Default setting: blank

Upgrade Rule

This parameter is a firmware upgrade script with the same syntax as Profile_Rule. Defines upgrade conditions and associated firmware URLs.
Default setting: blank

Log Upgrade Request Msg

Syslog message issued at the start of a firmware upgrade attempt.
Default setting: $PN $MAC -- Requesting upgrade $SCHEME://$SERVIP:$PORT$PATH

Log Upgrade Success Msg

Syslog message issued after a firmware upgrade attempt completes successfully.
Default setting: $PN $MAC -- Successful upgrade $SCHEME://$SERVIP:$PORT$PATH -- $ERR

Log Upgrade Failure Msg

Syslog message issued after a failed firmware upgrade attempt.
Default setting: $PN $MAC -- Upgrade failed: $ERR

License Keys

This field is not currently used.

Custom CA URL

The URL of a file location for a custom Certificate Authority (CA) certificate. Either the IP address or the FQDN of the server can be specified. The file name can have macros, such as $MA, which expands to the ATA MAC address.
Default setting: blank

GPP A to GPP P

General purpose provisioning parameters. These parameters can be used as variables in provisioning and upgrade rules. They are referenced by prepending the variable name with a ‘$’ character, such as $GPP_A.
Default setting: blank

Dial Tone

Prompts the user to enter a phone number. Reorder Tone is played automatically when Dial Tone or any of its alternatives times out.
Default setting: 350@-19,440@-19;10(*/0/1+2)

Second Dial Tone

Alternative to the Dial Tone when the user dials a three-way call.
Default setting: 420@-19,520@-19;10(*/0/1+2)

Outside Dial Tone

Alternative to the Dial Tone. It prompts the user to enter an external phone number, as opposed to an internal extension. It is triggered by a comma character encountered in the dial plan.
Default setting: 420@-16;10(*/0/1)

Prompt Tone

Prompts the user to enter a call forwarding phone number.
Default setting: 520@-19,620@-19;10(*/0/1+2)

Busy Tone

Played when a 486 RSC is received for an outbound call.
Default setting: 480@-19,620@-19;10(.5/.5/1+2)

Reorder Tone

Played when an outbound call has failed, or after the far end hangs up during an established call. Reorder Tone is played automatically when Dial Tone or any of its alternatives times out.
Default setting: 480@-19,620@-19;10(.25/.25/1+2)

Off Hook Warning Tone

Played when the caller has not properly placed the handset on the cradle. Off Hook Warning Tone is played when the Reorder Tone times out.
Default setting: 480@-10,620@0;10(.125/.125/1+2)

Ring Back Tone

Played during an outbound call when the far end is ringing.
Default setting: 440@-19,480@-19;*(2/4/1+2)

Ring Back 2 Tone

Your ATA plays this ringback tone instead of Ring Back Tone if the called party replies with a SIP 182 response without SDP to its outbound INVITE request.
Default setting: the same as Ring Back Tone, except the cadence is 1s on and 1s off.
Default setting: 440@-19,480@-19;*(1/1/1+2)

Confirm Tone

Brief tone to notify the user that the last input value has been accepted.
Default setting: 600@-16;1(.25/.25/1)

SIT1 Tone

Alternative to the Reorder Tone played when an error occurs as a caller makes an outbound call. The RSC to trigger this tone is configurable on the SIP screen.
Default setting: 985@-16,1428@-16,1777@-16;20(.380/0/1,.380/0/2,.380/0/3,0/4/0)

SIT2 Tone

Alternative to the Reorder Tone played when an error occurs as a caller makes an outbound call. The RSC to trigger this tone is configurable on the SIP screen.
Default setting: 914@-16,1371@-16,1777@-16;20(.274/0/1,.274/0/2,.380/0/3,0/4/0)

SIT3 Tone

Alternative to the Reorder Tone played when an error occurs as a caller makes an outbound call. The RSC to trigger this tone is configurable on the SIP screen.
Default setting: 914@-16,1371@-16,1777@-16;20(.380/0/1,.380/0/2,.380/0/3,0/4/0)

SIT4 Tone

Alternative to the Reorder Tone played when an error occurs as a caller makes an outbound call. The RSC to trigger this tone is configurable on the SIP screen.
Default setting: 985@-16,1371@-16,1777@-16;20(.380/0/1,.274/0/2,.380/0/3,0/4/0)

MWI Dial Tone

Played instead of the Dial Tone when there are unheard messages in the caller’s mailbox.
Default setting: 350@-19,440@-19;2(.1/.1/1+2);10(*/0/1+2)

Cfwd Dial Tone

Played when all calls are forwarded.
Default setting: 350@-19,440@-19;2(.2/.2/1+2);10(*/0/1+2)

Holding Tone

Informs the local caller that the far end has placed the call on hold.
Default setting: 600@-19;*(.1/.1/1,.1/.1/1,.1/9.5/1)

Conference Tone

Played to all parties when a three-way conference call is in progress.
Default setting: 350@-19;20(.1/.1/1,.1/9.7/1)

Secure Call Indication Tone

Played when a call has been successfully switched to secure mode. It should be played only for a short while (less than 30 seconds) and at a reduced level (less than -19 dBm) so it does not interfere with the conversation.
Default setting: 397@-19,507@-19;15(0/2/0,.2/.1/1,.1/2.1/2)

Feature Invocation Tone

Played when a feature is implemented.
Default setting: 350@-16;*(.1/.1/1)

Call Remind Tone

The holding tone is played on Phone ports to remind user of the presence of the held call during the active call when there are two calls.
Default setting: blank

Ring1 Cadence

Cadence script for distinctive ring 1.
Default setting: 60(2/4)

Ring2 Cadence

Cadence script for distinctive ring 2.
Default setting: 60(.8/.4,.8/4)

Ring3 Cadence

Cadence script for distinctive ring 3.
Default setting: 60(.4/.2,.4/.2,.8/4)

Ring4 Cadence

Cadence script for distinctive ring 4.
Default setting: 60(.3/.2,1/.2,.3/4)

Ring5 Cadence

Cadence script for distinctive ring 5.
Default setting: 1(.5/.5)

Ring6 Cadence

Cadence script for distinctive ring 6.
Default setting: 60(.2/.4,.2/.4,.2/4)

Ring7 Cadence

Cadence script for distinctive ring 7.
Default setting: 60(.4/.2,.4/.2,.4/4)

Ring8 Cadence

Cadence script for distinctive ring 8.
Default setting: 60(0.25/9.75)

CWT1 Cadence

Cadence script for distinctive CWT 1.
Default setting: *(.3/9.7)

CWT2 Cadence

Cadence script for distinctive CWT 2.
Default setting: 30(.1/.1,.1/9.7)

CWT3 Cadence

Cadence script for distinctive CWT 3.
Default setting: 30(.1/.1,.1/.1,.1/9.7)

CWT4 Cadence

Cadence script for distinctive CWT 4.
Default setting: 30(.1/.1,.3/.1,.1/9.3)

CWT5 Cadence

Cadence script for distinctive CWT 5.
Default setting: 1(.5/.5)

CWT6 Cadence

Cadence script for distinctive CWT 6.
Default setting: 30(.1/.1,.3/.2,.3/9.1)

CWT7 Cadence

Cadence script for distinctive CWT 7.
Default setting: 30(.3/.1,.3/.1,.1/9.1)

CWT8 Cadence

Cadence script for distinctive CWT 8.
Default setting: 2.3(.3/2)

Ring1 Name

Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 1 for the inbound call.
Default setting: Bellcore-r1

Ring2 Name

Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 2 for the inbound call.
Default setting: Bellcore-r2

Ring3 Name

Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 3 for the inbound call.
Default setting: Bellcore-r3

Ring4 Name

Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 4 for the inbound call.
Default setting: Bellcore-r4

Ring5 Name

Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 5 for the inbound call.
Default setting: Bellcore-r5

Ring6 Name

Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 6 for the inbound call.
Default setting: Bellcore-r6

Ring7 Name

Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 7 for the inbound call.
Default setting: Bellcore-r7

Ring8 Name

Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 8 for the inbound call.
Default setting: Bellcore-r8

Ring Waveform

Waveform for the ringing signal. Choices are Sinusoid or Trapezoid.
Default setting: Trapezoid

Ring Frequency

Frequency of the ringing signal. Valid values are 15–50 (Hz)
Default setting: 20

Ring Voltage

Ringing voltage. Choices are 30–90 (V)
Default setting: 85

CWT Frequency

Frequency script of the call waiting tone. All distinctive CWTs are based on this tone.
Default setting: 440@-10

Synchronized Ring

If this is set to yes, when the ATA is called, all lines ring at the same time (similar to a regular PSTN line) After one line answers, the others stop ringing.
Default setting: no

Hook Flash Timer Min

Minimum on-hook time before off-hook qualifies as hook flash. Less than this the on-hook event is ignored. Range: 0.1–0.4 seconds.
Default setting: 0.1

Hook Flash Timer Max

Maximum on-hook time before off-hook qualifies as hook flash. More than this the on-hook event is treated as on hook (no hook-flash event) Range: 0.4–1.6 seconds.
Default setting: 0.9

Callee On Hook Delay

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

Reorder Delay

Delay after far end hangs up before reorder tone is played. 0 = plays immediately, inf = never plays. Range: 0–255 seconds.
Default setting: 5.

Call Back Expires

Expiration time in seconds of a call back activation. Range: 0–65535 seconds.
Default setting: 1800

Call Back Retry Intvl

Call back retry interval in seconds. Range: 0–255 seconds.
Default setting: 30

Call Back Delay

Delay after receiving the first SIP 18x response before declaring the remote end is ringing. If a busy response is received during this time, the ATA still considers the call as failed and keeps on retrying.
Default setting: 0.5

VMWI Refresh Intvl

Interval between VMWI refresh to the device.
Default setting: 0

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 setting: 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 setting: 3

CPC Delay

Delay in seconds after caller hangs up when the ATA starts removing the tip-and-ring voltage to the attached equipment of the called party. The range is 0–255 seconds. This feature is generally used for answer supervision on the caller side to signal to the attached equipment when the call has been connected (remote end has answered) or disconnected (remote end has hung up) This feature should be disabled for the called party (in other words, by using the same polarity for connected and idle state) and the CPC feature should be used instead.

Without CPC enabled, reorder tone will is played after a configurable delay. If CPC is enabled, dial tone will be played when tip-to-ring voltage is restored. Resolution is 1 second.
Default setting: 2

CPC Duration

Duration in seconds for which the tip-to-ring voltage is removed after the caller hangs up. After that, tip-to-ring voltage is restored and the dial tone applies if the attached equipment is still off-hook. CPC is disabled if this value is set to 0. Range: 0 to 1.000 second. Resolution is 0.001 second.
Default setting: 0.5

Call Return Code

Call Return Code This code calls the last caller.
Default setting: *69

Call Redial Code

Redials the last number called.
Default setting: *07

Blind Transfer Code

Begins a blind transfer of the current call to the extension specified after the activation code.
Default setting: *98

Call Back Act Code

Starts a callback when the last outbound call is not busy.
Default setting: *66

Call Back Deact Code

Cancels a callback.
Default setting: *86

Call Back Busy Act Code

Starts a callback when the last outbound call is busy.
Default setting: *05

Cfwd All Act Code

Forwards all calls to the extension specified after the activation code.
Default setting: *72

Cfwd All Deact Code

Cancels call forwarding of all calls.
Default setting: *73

Cfwd Busy Act Code

Forwards busy calls to the extension specified after the activation code.
Default setting: *90

Cfwd Busy Deact Code

Cancels call forwarding of busy calls.
Default setting: *91

Cfwd No Ans Act Code

Forwards no-answer calls to the extension specified after the activation code.
Default setting: *92

Cfwd No Ans Deact Code

Cancels call forwarding of no-answer calls.
Default setting: *93

Cfwd Last Act Code

Forwards the last inbound or outbound call to the number that the user specifies after entering the activation code.
Default setting: *63

Cfwd Last Deact Code

Cancels call forwarding of the last inbound or outbound call.
Default setting: *83

Block Last Act Code

Blocks the last inbound call.
Default setting: *60

Block Last Deact Code

Cancels blocking of the last inbound call.
Default setting: *80

Accept Last Act Code

Accepts the last outbound call. It lets the call ring through when do not disturb or call forwarding of all calls are enabled.
Default setting: *64

Accept Last Deact Code

Cancels the code to accept the last outbound call.
Default setting: *84

CW Act Code

Enables call waiting on all calls.
Default setting: *56

CW Deact Code

Disables call waiting on all calls.
Default setting: *57

CW Per Call Act Code

Enables call waiting for the next call.
Default setting: *71

CW Per Call Deact Code

Disables call waiting for the next call.
Default setting: *70

Block CID Act Code

Blocks caller ID on all outbound calls.
Default setting: *67

Block CID Deact Code

Removes caller ID blocking on all outbound calls.
Default setting: *68

Block CID Per Call Act Code

Blocks caller ID on the next outbound call.
Default setting: *81

Block CID Per Call Deact Code

Removes caller ID blocking on the next inbound call.
Default setting: *82

Block ANC Act Code

Blocks all anonymous calls.
Default setting: *77

Block ANC Deact Code

Removes blocking of all anonymous calls.
Default setting: *87

DND Act Code

Enables the do not disturb feature.
Default setting: *78

DND Deact Code

Disables the do not disturb feature.
Default setting: *79

CID Act Code

Enables caller ID generation.
Default setting: *65

CID Deact Code

Disables caller ID generation.
Default setting: *85

CWCID Act Code

Enables call waiting, caller ID generation.
Default setting: *25

CWCID Deact Code

Disables call waiting, caller ID generation.
Default setting: *45

Dist Ring Act Code

Enables the distinctive ringing feature.
Default setting: *26

Dist Ring Deact Code

Disables the distinctive ringing feature.
Default setting: *46

Speed Dial Act Code

Assigns a speed dial number.
Default setting: *74

Paging Code

Used for paging other clients in the group.
Default setting: *96

Secure All Call Act Code

Makes all outbound calls secure.
Default setting: *16

Secure No Call Act Code

Makes all outbound calls not secure.
Default setting: *17

Secure One Call Act Code

Makes the next outbound call secure. (It is redundant if all outbound calls are secure by default.)
Default setting: *18

Secure One Call Deact Code

Makes the next outbound call not secure. (It is redundant if all outbound calls are not secure by default.)
Default setting: *19

Conference Act Code

If this code is specified, the user must enter it before dialing the third party for a conference call. Enter the code for a conference call.
Default setting: blank

Attn-Xfer Act Code

If the code is specified, the user must enter it before dialing the third party for a call transfer. Enter the code for a call transfer.
Default setting: blank

Modem Line Toggle Code

Toggles the line to a modem. Modem passthrough mode can be triggered only by pre-dialing this code.
Default setting: *99

FAX Line Toggle Code

Toggles the line to a fax machine.
Default setting: #99

Media Loopback Code

Use for media loopback.
Default setting: *03

Referral Services Codes

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

One or more *codes can be configured into this parameter, such as *98, or *97|*98|*123, etc. The maximum length is 79 characters. This parameter applies when the user places the current call on hold by pressing the hook flash button. Each *code (and the following valid target number according to current dial plan) triggers the ATA to perform a blind transfer to a target number that is prepended by the service *code.

For example, after the user dials *98, the ATA 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 ATA sends a blind REFER to the holding party with the Refer-To target equal to *98 target_number. This feature allows the ATA 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 ATA. You can empty the corresponding *code that you do not want the ATA to process.
Default setting: blank

Feature Dial Services Codes

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

One or more *codes can be configured into this parameter, such as *72, or *72|*74|*67|*82, etc. The maximum length is 79 characters. This parameter applies when the user has a dial tone (first or second dial tone) After receiving dial tone, a user enters the *code and the target number according to current dial plan. For example, after user dials *72, the ATA plays a special tone called a Prompt tone while awaiting the user to enter a valid target number. When a complete number is entered, the ATA 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 ATA. You can remove a corresponding *code that you do not want to the ATA to process.

You can add a parameter to indicate which tone plays after the *code is entered, such as *72‘c‘|*67‘p‘. Below is a list of allowed tone parameters (note the use of open 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 ATA 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 this parameter. Instead, add the *code in the dial plan and the ATA send INVITE *73@..... as usual when user dials *73.
Default setting: blank

Service Annc Base Number

Base number for service announcements.
Default setting: blank

Service Annc Extension Codes

Extension codes for service announcements.
Default setting: blank

Prefer G711u Code

Dial prefix to make G.711u the preferred codec for the call.
Default setting: *017110

Force G711u Code

Dial prefix to make G.711u the only codec that can be used for the call.
Default setting: *027110

Prefer G711a Code

Dial prefix to make G.711a the preferred codec for the call.
Default setting: *017111

Force G711a Code

Dial prefix to make G.711a the only codec that can be used for the call.
Default setting: *027111

Prefer G726r32 Code

Dial prefix to make G.726r32 the preferred codec for the call.
Default setting: *0172632

Force G726r32 Code

Dial prefix to make G.726r32 the only codec that can be used for the call.
Default setting: *0272632

Prefer G729a Code

Dial prefix to make G.729a the preferred codec for the call.
Default setting: *01729

Force G729a Code

Dial prefix to make G.729a the only codec that can be used for the call.
Default setting: *02729

FXS Port Impedance:

Sets the electrical impedance of the PHONE port.

Choices are:

• 600
• 900
• 600+2.16μF
• 900+2.16μF
• 220+850||120nF
• 220+820||115nF
• 200+600||100nF

Default setting: 600.

FXS Port Input Gain:

Input gain in dB, up to three decimal places. The range is 6.000 to -12.000.

Default setting: -3.

FXS Port Output Gain:

Output gain in dB, up to three decimal places. The range is 6.000 to -12.000. The Call Progress Tones and DTMF playback level are not affected by the FXS Port Output Gain parameter.

Default setting: -3.

DTMF Playback Level:

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

Default setting: -16.0.

DTMF Twist:

To gain difference between the two tone frequency.

Default setting: 2

DTMF Playback Length:

Local DTMF playback duration in milliseconds.

Default setting:.1.

Detect ABCD:

To enable local detection of DTMF ABCD, select Yes. Otherwise, select No. Default setting: Yes

This setting has no effect if DTMF Tx Method is INFO; ABCD is always sent OOB regardless in this setting.

Playback ABCD:

To enable local playback of OOB DTMF ABCD, select Yes. Otherwise, select No. Default setting: Yes

Caller ID Method:

Your choices are:

• Bellcore (N.Amer,China): CID, CIDCW, and VMWI. FSK sent after first ring (same as ETSI FSK sent after first ring) (no polarity reversal or DTAS).
• DTMF (Finland, Sweden): CID only. DTMF sent after polarity reversal (and no DTAS) and before first ring.
• DTMF (Denmark): CID only. DTMF sent before first ring with no polarity reversal and no DTAS.
• ETSI DTMF: CID only. DTMF sent after DTAS (and no polarity reversal) and before first ring.
• ETSI DTMF With PR: CID only. DTMF sent after polarity reversal and DTAS and before first ring.
• ETSI DTMF After Ring: CID only. DTMF sent after first ring (no polarity reversal or DTAS).
• ETSI FSK: CID, CIDCW, and VMWI. FSK sent after DTAS (but no polarity reversal) and before first ring. Waits for ACK from a device after DTAS for CIDCW.
• ETSI FSK With PR (UK): CID, CIDCW, and VMWI. FSK is sent after polarity reversal and DTAS and before first ring. Waits for ACK from a device after DTAS for CIDCW. Polarity reversal is applied only if equipment is on hook.
• DTMF (Denmark) with PR: CID only. DTMF sent after polarity reversal (and no DTAS) and before first ring.

Default setting: Bellcore(N.Amer, China)

FXS Port Power Limit:

The choices are from 1 to 8. Default setting: 3

Caller ID FSK Standard:

The ATA supports bell 202 and v.23 standards for caller ID generation. Default setting: bell 202

Feature Invocation Method:

Select the method you want to use, Default, or Sweden default. Default setting: Default.

Line Enable

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

SAS Enable

To enable the use of the line as a streaming audio source, select yes. Otherwise, select no. If enabled, the line cannot be used for outgoing calls. Instead, it auto-answers incoming calls and streams audio RTP packets to the caller.
Default setting: no

SAS DLG Refresh Intvl

If this value is not zero, it is the interval at which the streaming audio server sends out session refresh (SIP re-INVITE) messages to determine whether the connection to the caller is still active. If the caller does not respond to the refresh message, the ATA ends this call with a SIP BYE message. The range is 0 to 255 seconds (0 means that the session refresh is disabled)
Default setting: 30

SAS Inbound RTP Sink

The purpose of this parameter is to work around devices that do not play inbound RTP if the SAS line declares itself as a send-only device and tells the client not to stream out audio. This parameter is an FQDN or IP address of an RTP sink to be used by the SAS line in the SDP of its 200 response to inbound INVITE from a client. It will appear in the c = line and the port number, if specified, will appear in the m = line of the SDP. If this value is not specified or is equal to 0, then c = 0.0.0.0 and a=sendonly will be used in the SDP to tell the SAS client to not to send any RTP to this SAS line. If a non-zero value is specified, then a=sendrecv and the SAS client will stream audio to the given address. Special case: If the value is $IP, then the SAS line’s own IP address is used in the c = line and a=sendrecv. In that case the SAS client will stream RTP packets to the SAS line.
Default setting: blank

NAT Mapping Enable

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

NAT Keep Alive Enable

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

NAT Keep Alive Msg

Enter the keep alive message that should be sent periodically to maintain the current NAT mapping. If the value is $NOTIFY, a NOTIFY message is sent. If the value is $REGISTER, a REGISTER message without contact is sent.
Default setting: $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 proxy server or outbound proxy server.
Default setting: $PROXY

SIP ToS/DiffServ Value

TOS/DiffServ field value in UDP IP packets carrying a SIP message.
Default setting: 0x68

SIP CoS Value [0-7]

CoS value for SIP messages. Valid values are 0 through 7.
Default setting: 3

RTP ToS/DiffServ Value

ToS/DiffServ field value in UDP IP packets carrying RTP data.
Default setting: 0xb8

RTP CoS Value [0- 7]

CoS value for RTP data. Valid values are 0 through 7.
Default setting: 6

Network Jitter Level

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

Jitter Buffer Adjustment

Choose yes to enable or no to disable this feature.
Default setting: yes

SIP Transport

The TCP choice provides “guaranteed delivery”, which assures that lost packets are retransmitted. TCP also guarantees that the SIP packages are received in the same order that they were sent. As a result, TCP overcomes the main disadvantages of UDP. In addition, for security reasons, most corporate firewalls block UDP ports. With TCP, new ports do not need to be opened or packets dropped, because TCP is already in use for basic activities such as Internet browsing or e-commerce. Options are: UDP, TCP, TLS. Default setting: UDP

SIP Port

Port number of the SIP message listening and transmission port.
Default setting: 5060 for PHONE1 and 5061 for PHONE2

SIP 100REL Enable

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

EXT SIP Port

The external SIP port number.
Default setting: blank

Auth Resync-Reboot

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

SIP Proxy-Require

The SIP proxy can support a specific extension or behavior when it sees this header from the user agent. 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.
Default setting: blank

SIP Remote-Party-ID

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

SIP GUID

This feature limits the registration of SIP accounts. The Global Unique ID is generated for each line for each ATA. When it is enabled, the ATA adds a GUID header in the SIP request. The GUID is generated the first time the unit boots up and stays with the unit through rebooting and even factory reset.
Default setting: no

SIP Debug Option

SIP messages are received at or sent from the proxy listen port. This feature controls which SIP messages to log. The choices are described below. Default setting: none

• none— No logging.
• 1-line— Logs the start-line only for all messages.
• 1-line excl. OPT— Logs the start-line only for all messages except OPTIONS requests/responses.
• 1-line excl. NTFY— Logs the start-line only for all messages except NOTIFY requests/responses.
• 1-line excl. REG— Logs the start-line only for all messages except REGISTER requests/responses.
• 1-line excl. OPT|NTFY|REG— Logs the start-line only for all messages except OPTIONS, NOTIFY, and REGISTER requests/responses.
• full— Logs all SIP messages in full text.
• full excl. OPT— Logs all SIP messages in full text except OPTIONS requests/responses.
• full excl. NTFY— Logs all SIP messages in full text except NOTIFY requests/responses.
• full excl. REG— Logs all SIP messages in full text except REGISTER requests/responses.
• full excl. OPT|NTFY|REG— Logs all SIP messages in full text except for OPTIONS, NOTIFY, and REGISTER requests/responses.

RTP Log Intvl

The interval for the RTP log.
Default setting: 0

Restrict Source IP

If configured, the ATA drops all packets sent to its SIP Ports from an untrusted IP address. A source IP address is untrusted if it does not match any of the IP addresses resolved from the configured Proxy (or Outbound Proxy if Use Outbound Proxy is yes)
Default setting: no

Referor Bye Delay

The number of seconds to wait before sending a BYE to the referrer to terminate a stale call leg after a call transfer.)
Default setting: 4

Refer Target Bye Delay

The number of seconds to wait before sending a BYE to the refer target to terminate a stale call leg after a call transfer.)
Default setting: 0

Referee Bye Delay

The number of seconds to wait before sending a BYE to the referee to terminate a stale call leg after a call transfer.)
Default setting: 0

Refer-To Target Contact

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

Sticky 183

If this feature is enabled, the ATA 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 setting: no

Auth INVITE

When enabled, authorization is required for initial incoming INVITE requests from the SIP proxy.
Default setting: no

Reply 182 On Call Waiting

When enabled, the ATA replies with a SIP182 response to the caller if it is already in a call and the line is off-hook. To use this feature select yes.
Default setting: no

Use Anonymous With RPID

Determines whether or not the ATA uses “Anonymous” when Remote Party ID is requested in the SIP message.
Default setting: yes

Use Local Addr In From

Use the local ATA IP address in the SIP FROM message.
Default setting: no

Blind Attn-Xfer Enable

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

MOH Server

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

Xfer When Hangup Conf

Makes the ATA perform a transfer when a conference call has ended. Select yes or no from the drop-down menu.
Default setting: yes

Conference Bridge URL

This feature supports external conference bridging for n-way conference calls (n>2), instead of mixing audio locally. To use this feature, set this parameter to that of the server's name. For example: conf@mysefver.com:12345 or conf (which uses the Proxy value as the domain).
Default setting: blank

Conference Bridge Ports

Select the maximum number of conference call participants. The range is 3 to 10.
Default setting: 3

Enable IP Dialing

Enable or disable IP dialing. If IP dialing is enabled, one can dial [userid@] a.b.c.d[:port], where ‘@’, ‘.’, and ‘:’ are dialed by entering *, user-id must be numeric (like a phone number) and a, b, c, d must be between 0 and 255, and port must be larger than 255. If port is not given, 5060 is used. Port and User-Id are optional. If the user-id portion matches a pattern in the dial plan, then it is interpreted as a regular phone number according to the dial plan. The INVITE message, however, is still sent to the outbound proxy if it is enabled.
Default setting: no

Emergency Number

Comma separated list of emergency number patterns. If outbound call matches one of the pattern, the ATA will disable hook flash event handling. The condition is restored to normal after the call ends. Blank signifies that there is no emergency number. Maximum number length is 63 characters.
Default setting: blank

Mailbox ID

Enter the ID number of the mailbox for this line.
Default setting: blank

Feature Key Sync

Allows the phone to synchronize with the call server. If yes is selected, if Do Not Disturb or Call Forwarding settings are changed on the phone, changes are also made on the server; if changes are made on the server, they are propagated to the phone.
Default setting: no

Proxy

SIP proxy server for all outbound requests.
Default setting: blank

Outbound Proxy

SIP Outbound Proxy Server where all outbound requests are sent as the first hop.
Default setting: blank

Use Outbound Proxy

Enables the use of an Outbound Proxy. If set to no, the Outbound Proxy and Use OB Proxy in Dialog parameters are ignored.
Default setting: no

Use OB Proxy In Dialog

Whether to force SIP requests to be sent to the outbound proxy within a dialog. Ignored if the parameter Use Outbound Proxy is no, or the Outbound Proxy parameter is empty.
Default setting: yes

Register

Enable periodic registration with the Proxy parameter. This parameter is ignored if Proxy is not specified.
Default setting: yes

Make Call Without Reg

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

Register Expires

Expires value in sec in a REGISTER request. The ATA will periodically renew registration shortly before the current registration expired. This parameter is ignored if the Register parameter is no. Range: 0 – (231 – 1) sec.
Default setting: 3600

Ans Call Without Reg

Allow answering inbound calls without successful (dynamic) registration by the unit.
Default setting: no

Use DNS SRV

Whether to use DNS SRV lookup for Proxy and Outbound Proxy.
Default setting: no

DNS SRV Auto Prefix

If enabled, the ATA will automatically prepend the Proxy or Outbound Proxy name with _sip._udp when performing a DNS SRV lookup on that name.
Default setting: no

Proxy Fallback Intvl

After failing over to a lower priority server, the ATA waits for the specified Proxy Fallback Interval, in seconds, before retrying the highest priority proxy (or outbound proxy) servers. This parameter is useful only if the primary and backup proxy server list is provided to the ATA via DNS SRV record lookup on the server name. (Using multiple DNS A records per server name does not allow the notion of priority, so all hosts will be considered at the same priority and the ATA will not attempt to fall back after a failover.)
Default setting: 3600

Proxy Redundancy Method

The method that the ATA uses to create a list of proxies returned in the DNS SRV records. If you select Normal, the list will contain proxies ranked by weight and priority. If you select Based on SRV port, the ATA also inspects the port number based on 1st proxy’s port.
Default setting: Normal

Mailbox Subscribe URL

The URL or IP address of the voicemail server.
Default setting: blank

Mailbox Subscribe Expires

Sets subscription interval for voicemail message waiting indication. When this time period expires, the ATA sends another subscribe message to the voice mail server. Default: 2147483647

Auto Register when Failover

This parameter controls the failover behavior when there is an error.
Default setting: no.

There are two options:

• No - Resend directly to backup proxy.
• Yes - Unregister with current proxy, register to backup proxy and then resend to backup proxy.

This option is used under the following conditions:

–The failed message is an INVITE message.

–The return code matches one of the values in parameter Try Backup RSC.

Display Name

Display name for caller ID.
Default setting: blank

User ID

User ID for this line.
Default setting: blank

Password

Password for this line.
Default setting: blank

Use Auth ID

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

Auth ID

Authentication ID for SIP authentication.
Default setting: blank

Resident Online Number

This setting allows you to associate a "local" telephone number with this line using a valid Skype Online Number from Skype. Calls made to that number will ring your phone. Enter the number without spaces or special characters.
Default setting: 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 that contains no @ character, then the actual URI used in the SIP signaling should be automatically formed by appending this parameter with an @ character followed by the IP address of the device.

Call Waiting Serv

Enable Call Waiting Service.
Default setting: yes

Block CID Serv

Enable Block Caller ID Service.
Default setting: yes

Block ANC Serv

Enable Block Anonymous Calls Service
Default setting: yes

Dist Ring Serv

Enable Distinctive Ringing Service
Default setting: yes

Cfwd All Serv

Enable Call Forward All Service
Default setting: yes

Cfwd Busy Serv

Enable Call Forward Busy Service
Default setting: yes

Cfwd No Ans Serv

Enable Call Forward No Answer Service
Default setting: yes

Cfwd Sel Serv

Enable Call Forward Selective Service. Configure this service in the Selective Call Forward Settings section.
Default setting: yes

Cfwd Last Serv

Enable Forward Last Call Service
Default setting: yes

Block Last Serv

Enable Block Last Call Service
Default setting: yes

Accept Last Serv

Enable Accept Last Call Service
Default setting: yes

DND Serv

Enable Do Not Disturb Service
Default setting: yes

CID–Serv

Enable Caller ID Service
Default setting: yes

CWCID Serv

Enable Call Waiting Caller ID Service
Default setting: yes

Call Return Serv

Enable Call Return Service
Default setting: yes

Call Redial Serv

Enable Call Redial Service.
Default setting: yes

Call Back Serv

Enable Call Back Service.
Default setting: yes

Three Way Call Serv

Enable Three Way Calling Service. Three Way Calling is required for Three Way Conference and Attended Transfer.
Default setting: yes

Three Way Conf Serv

Enable Three Way Conference Service. Three Way Conference is required for Attended Transfer.
Default setting: yes

Attn Transfer Serv

Enable Attended Call Transfer Service. Three Way Conference is required for Attended Transfer.
Default setting: yes

Unattn Transfer Serv

Enable Unattended (Blind) Call Transfer Service.
Default setting: yes

MWI Serv

Enable MWI Service. MWI is available only if a Voice Mail Service is set-up in the deployment.
Default setting: yes

VMWI Serv

Enable VMWI Service (FSK)
Default setting: yes

Speed Dial Serv

Enable Speed Dial Service.
Default setting: yes

Secure Call Serv

Secure Call Service. If this feature is enabled, a user can make a secure call by entering an activation code (*18 by default) before dialing the target number. Then audio traffic in both directions is encrypted for the duration of the call.
Default setting: yes

Star codes are set in Vertical Service Activation Codes. To enable secure calling by default, without requiring a star code, set the user’s Secure Call Setting to yes. See User 1 and User 2.

Referral Serv

Enable Referral Service. See the Referral Services Codes parameter For more information.
Default setting: yes

Feature Dial Serv

Enable Feature Dial Service. See the Feature Dial Services Codes parameter For more information.
Default setting: yes

Service Announcement Serv

Enable Service Announcement Service.
Default setting: no

Reuse CID Number As Name

Use the Caller ID number as the caller name.
Default settings: yes

CONFCID Serv

Enable Caller ID during conference call.
Default settings: yes

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
• G726-32
• G729a

Default setting: G711u.

Second Preferred Codec

If the first codec fails, then second preferred codec is tried.

Default setting: blank

Third Preferred Codec

If the second codec fails, then third preferred codec is tried.

Default setting: blank

Use Pref Codec Only

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

Use Remote Pref Codec

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

Codec Negotiation

When set to Default, the Cisco IP phone responds to an Invite with a 200 OK response advertising the preferred codec only. When set to List All, the Cisco IP phone responds listing all the codecs that the phone supports.
Default setting: Default

G729a Enable

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

Silence Supp Enable

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

G726-32 Enable

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

Silence Threshold

Select the appropriate setting for the threshold: high, medium, or low.
Default setting: medium

FAX V21 Detect Enable

To enable detection of V21 fax tones, select yes. Otherwise, select no.
Default setting: yes

Echo Canc Enable

To enable the use of the echo canceller, select yes. Otherwise, select no.
Default setting: yes

FAX CNG Detect Enable

To enable detection of the fax Calling Tone (CNG), select yes. Otherwise, select no.
Default setting: yes

FAX Passthru Codec

Select the codec for fax passthrough, G711u or G711a.
Default setting: G711u

FAX Codec Symmetric

To force the ATA to use a symmetric codec during fax passthrough, select yes. Otherwise, select no.
Default setting: yes

DTMF Process INFO

To use the DTMF process info feature, select yes. Otherwise, select no.
Default setting: yes

FAX Passthru Method

Select the fax passthrough method: None, NSE, or ReINVITE.
Default setting: NSE

DTMF Process AVT

To use the DTMF process AVT feature, select yes. Otherwise, select no.
Default setting: yes

FAX Process NSE

To use the fax process NSE feature, select yes. Otherwise, select no.
Default setting: yes

DTMF Tx Method

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

FAX Disable ECAN

If enabled, this feature automatically disables the echo canceller when a fax tone is detected. To use this feature, select yes. Otherwise, select no.
Default setting: no

DTMF Tx Mode

DTMF Detection Tx Mode is available for SIP information and AVT. Options are: Strict or Normal.
Default setting: Strict for which the following are true:

• A DTMF digit requires an extra hold time after detection.
• The DTMF level threshold is raised to -20 dBm.

The minimum and maximum duration thresholds are:

• strict mode for AVT and SIP: the value set in DTMF Tx Strict Hold Off Time
• normal mode for AVT: 40 ms
• normal mode for SIP: 50 ms

DTMF Tx Strict Hold Off Time

This parameter is in effect only when DTMF Tx Mode is set to strict, and when DTMF Tx Method is not set to inband; i.e. either AVT or INFO. The value can be set as low as 40 ms. There is no maximum limit. A larger value will reduce the chance of talk-off (beeping) during conversation, at the expense of reduced performance of DTMF detection, which is needed for interactive voice response systems (IVR).
Default setting: 70 ms

FAX Enable T38

To enable the use of ITU-T T.38 standard for FAX Relay, select yes. Otherwise select no.
Default setting: no

Hook Flash Tx Method

Select the method for signaling hook flash events: None, AVT, or INFO. None does not signal hook flash events. AVT uses RFC2833 AVT (event = 16) INFO uses SIP INFO with the single line signal=hf in the message body. The MIME type for this message body is taken from the Hook Flash MIME Type setting.
Default setting: None

FAX T38 Redundancy

Select the appropriate number to indicate the number of previous packet payloads to repeat with each packet. Choose 0 for no payload redundancy. The higher the number, the larger the packet size and the more bandwidth consumed.
Default setting: 1

FAX T38 ECM Enable

Select yes to enable T.38 Error Correction Mode. Otherwise select no.
Default setting: yes

FAX Tone Detect Mode

This parameter has three possible values:

• caller or callee : The ATA will detect FAX tone whether it is callee or caller
• caller only: The ATA will detect FAX tone only if it is the caller
• callee only : The ATA will detect FAX tone only if it is the callee
  Default setting: caller or callee.

Symmetric RTP

Enable symmetric RTP operation. If enabled, the ATA sends RTP packets to the source address and port of the last received valid inbound RTP packet. If disabled (or before the first RTP packet arrives) the ATA sends RTP to the destination as indicated in the inbound SDP.
Default setting: no

Fax T38 Return to Voice

When this feature is enabled, upon completion of the fax image transfer, the connection remains established and reverts to a voice call using the previously designated codec. Select yes to enable this feature, or select no to disable it.
Default setting: no

Modem Line

Enable an alternate method to make the modem call without Modem Line Toggle Code pre-dialing.
Default setting: no

RTP to Proxy in Remote Hold

Enable to send RTP to proxy when line is held by remote side.
Default setting: no

*xx

Allow arbitrary 2 digit star code

[3469]11

Allow x11 sequences

0

Operator

00

Int’l Operator

[2-9]xxxxxx

US local number

1xxx[2-9]xxxxxx

US 1 + 10-digit long distance number

xxxxxxxxxxxx.

Everything else

Idle Polarity

Polarity before a call is connected: Forward or Reverse.
Default setting: Forward

Caller Conn Polarity

Polarity after an outbound call is connected: Forward or Reverse.
Default setting: Forward.

Callee Conn Polarity

Polarity after an inbound call is connected: Forward or Reverse.
Default setting: Forward

Cfwd All Dest

Forward number for Call Forward All Service.
Default setting: blank

Cfwd Busy Dest

Forward number for Call Forward Busy Service. Same as Cfwd All Dest.
Default setting: blank

Cfwd No Ans Dest

Forward number for Call Forward No Answer Service. Same as Cfwd All Dest.
Default setting: blank

Cfwd No Ans Delay

Delay in sec before Call Forward No Answer triggers.
Default setting: 20