Cisco WRP500 Administration Guide

Product Name

Model number/name.

Serial Number

Serial number.

Software Version

Software version number.

Hardware Version

Hardware version number.

MAC Address

MAC address.

Client Certificate

Status of the client certificate, which can indicate whether the WRP500 has been authorized by your ITSP.

Customization

For a Remote Configuration (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.

Voice Module Version

Voice module number.

Current Time

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

Elapsed Time

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

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

External IP address used for NAT mapping.

Hook State

Hook state of the FXS port. Options are either 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

Number of seconds before the next registration renewal.

Message Waiting

Indicates whether you have new voice mail waiting. Options are either Yes or No. The value automatically is set to Yes when a message is received. You also can clear or set the flag manually. Setting this value to Yes can activate stutter tone and VMWI signal. This parameter is stored in long term memory and survives after reboot or power cycle.

Call Back Active

Indicates whether a call back request is in progress. Options are either Yes or No.

Last Called Number

The last number called from the FXS line.

Last Caller Number

Number of the last caller.

Mapped SIP Port

Port number of the SIP port mapped by NAT.

Call 1 and 2 State

May take one of the following values:

• Idle
• Dialing
• Stunning
• Calling
• Proceeding
• Ringing
• Invalid
• Connected
• Hold
• Holding
• Resuming
• Transit

Call 1 and 2 Tone

Type of tone used by the call.

Call 1 and 2 Encoder

Codec used for encoding.

Call 1 and 2 Decoder

Codec used for decoding.

Call 1 and 2 FAX

Status of the fax mode.

Call 1 and 2 Type

Direction of the call. May take one of the following values:

• Inbound
• Outbound
• Transferred

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

Name of the internal phone.

Call 1 and 2 Peer Phone

Phone number of the internal phone.

Call 1 and 2 Call Duration

Duration of the call.

Call 1 and 2 Packets Sent

Number of packets sent.

Call 1 and 2 Packets Recv

Number of packets received.

Call 1 and 2 Bytes Sent

Number of bytes sent.

Call 1 and 2 Bytes Recv

Number of bytes received.

Call 1 and 2 Decode Latency

Number of milliseconds for decoder latency.

Call 1 and 2 Jitter

Number of milliseconds for receiver jitter.

Call 1 and 2 Packets Lost

Number of packets lost.

Call 1 and 2 Packet Error

Number of invalid packets received.

Call 1 and 2 Mapped RTP Port

The port mapped for Real Time Protocol traffic for Call 1/2.

Call 1 and 2 Media Loopback

Media loopback is used to quantitatively and qualitatively measure the voice quality that the end user experiences.

Restricted Access Domains

This feature is used when implementing software customization.

IVR Admin Passwd

Password for entering IVR menu.

Syslog Server

Specifies the IP address of the syslog server.

Debug Server

Specifies the IP address of the debug server, which logs debug information. The level of detailed output depends on the debug level parameter setting.

Debug Level

Determines the level of debug information that is generated. Select 0, 1, 2, or 3 from the drop-down menu. The higher the debug level, the more debug information is generated.

The default is 0, which indicates that no debug information is generated.

Debug Option

Specifies what debug information is expected. Generally can be set to dbg_all.

Max Forward

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

The default is 70.

Max Redirection

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

The default is 5.

Max Auth

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

The default is 2.

SIP User Agent Name

User-Agent header used in outbound requests.

The default is $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.

The default is $VERSION.

SIP Reg User Agent Name

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.

The default is blank.

SIP Accept Language

Accept-Language header used. There is no default (this indicates the WRP500 does not include this header). If empty, the header is not included.

DTMF Relay MIME Type

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

The default is application/dtmf-relay.

Remove Last Reg

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

The default is no.

Use Compact Header

Lets you use compact SIP headers in outbound SIP messages. Select yes or no from the drop-down menu. If set to yes, the WRP500 uses compact SIP headers in outbound SIP messages. If set to no, the WRP500 uses normal SIP headers. If inbound SIP requests contain compact headers, the WRP500 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 WRP500 substitutes those headers with compact headers (if defined by RFC 261) if Use Compact Header parameter is set to yes.

The default is no.

Escape Display Name

Lets you keep the Display Name private. Select yes if you want the WRP500 to enclose the string (configured in the Display Name) in a pair of double quotes for outbound SIP messages. Any occurrences of or \ in the string is escaped with \ and \\ inside the pair of double quotes. Otherwise, select no.

The default is no.

RFC 2543 Call Hold

Configures the type of call hold: a:sendonly or 0.0.0.0.

The default is no; do not use the 0.0.0.0 syntax in a HOLD SDP; use the a:sendonly syntax.

Mark All AVT Packets

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

The default is yes.

SIP TCP Port Min

Specifies the lowest TCP port number that can be used for SIP sessions. The default Port Min is 5060.

SIP TCP Port Max

Specifies the highest TCP port number that can be used for SIP sessions. The default Port Max is 5080.

SIP T1

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

The default is 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.

The default is 4.

SIP T4

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

The default is 5.

SIP Timer B

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

The default is 32.

SIP Timer F

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

The default is 32.

SIP Timer H

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

The default is 32.

SIP Timer D

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

The default is 32.

SIP Timer J

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

The default is 32.

INVITE Expires

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

The default is 240. Range: 0–(2 ³¹ –1).

ReINVITE Expires

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

The default is 30. Range: 0–(2 ³¹ –1).

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.

The default is 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.

The default is 7200.

Reg Retry Intvl

Interval to wait before the WRP500 retries registration after failing during the last registration.

The default is 30.

Reg Retry Long Intvl

When registration fails with a SIP response code that does not match
Retry Reg RSC, the WRP500 waits for the specified length of time before retrying. If this interval is 0, the WRP500 stops trying. This value should be much larger than the Reg Retry Intvl value, which should not be 0.

The default is 1200.

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.

SIT2 RSC

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

SIT3 RSC

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

SIT4 RSC

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

Try Backup RSC

SIP response code that retries a backup server for the current request.

Retry Reg RSC

Interval to wait before the WRP500 retries registration after failing during the last registration.

The default is 30.

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.

The default is 16384.

RTP Port Max

Maximum port number for RTP transmission and reception.

The default is 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.

The default is 0.030.

Stats In BYE

Determines whether the WRP500 includes the P-RTP-Stat header or response to a BYE message. The header contains the RTP statistics of the current call. Select yes or no from the drop-down menu. The format of the P-RTP-Stat header is:

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

The default is no.

NSE Dynamic Payload

NSE dynamic payload type. The valid range is 96-127.

The default is 100.

AVT Dynamic Payload

AVT dynamic payload type. The valid range is 96-127.

The default is 101.

INFOREQ Dynamic Payload

INFOREQ dynamic payload type.

There is no default.

NSE Codec Name

NSE codec name used in SDP.

The default is NSE.

AVT Codec Name

AVT codec name used in SDP.

The default is telephone-event.

G711u Codec Name

G.711u codec name used in SDP.

The default is PCMU.

G711a Codec Name

G.711a codec name used in SDP.

The default is PCMA.

G729a Codec Name

G.729a codec name used in SDP.

The default is G729a.

G729b Codec Name

G.729b codec name used in SDP.

The default is G729ab.

EncapRTP Codec Name

EncapRTP codec name used in SDP.

The default is EncapRTP.

EncapRTP Dynamic Payload

EncapRTP dynamic payload type.

Handle VIA received

If you select yes, the WRP500 processes the received parameter in the VIA header (this value is inserted by the server in a response to anyone of its requests). If you select no, the parameter is ignored. Select yes or no from the drop-down menu.

The default is no.

Handle VIA rport

If you select yes, the WRP500 processes the rport parameter in the VIA header (this value is inserted by the server in a response to anyone of its requests). If you select no, the parameter is ignored. Select yes or no from the drop-down menu.

The default is 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.

The default is 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.

The default is 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.

The default is 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.

The default is no.

STUN Enable

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

The default is no.

STUN Test Enable

If the STUN Enable feature is enabled and a valid STUN server is available, the WRP500 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 WRP500 detects symmetric NAT or a symmetric firewall, NAT mapping is disabled.

The default is no.

STUN Server

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

EXT IP

External IP address to substitute for the actual IP address of the WRP500 in all outgoing SIP messages. If 0.0.0.0 is specified, no IP address substitution is performed.

If this parameter is specified, the WRP500 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.

Note 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 WRP500 is the edge device, the second requirement is met.

The default is 0.0.0.0.

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.

The default is 0.

NAT Keep Alive Intvl

Interval between NAT-mapping keep alive messages.

The default is 15.

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.

The default is 350@-19,440@-19;10(*/0/1+2).

Second Dial Tone

Alternative to the Dial Tone when the user dials a three-way call.

The default is 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.

The default is 420@-19;10(*/0/1).

Prompt Tone

Prompts the user to enter a call forwarding phone number.

The default is 520@-19,620@-19;10(*/0/1+2).

Busy Tone

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

The default is 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.

The default is 480@-19,620@-19;10(.25/.25/1+2).

Off Hook Warning Tone

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

The default is 480@10,620@0;10(.125/.125/1+2)

Ring Back Tone

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

The default is 440@-19,480@-19;*(2/4/1+2).

Ring Back 2 Tone

Your WRP500 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. The default value is the same as Ring Back Tone, except the cadence is 1s on and 1s off.

The default is 440@-19,480@-19;*(1/1/1+2).

Confirm Tone

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

The default is 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.

The default is 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.

The default is 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.

The default is 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.

The default is 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.

The default is 350@-19,440@-19;2(.1/.1/1+2);10(*/0/1+2).

Cfwd Dial Tone

Played when all calls are forwarded.

The default is 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.

The default is 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.

The default is 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.

The default is 397@-19,507@-19;15(0/2/0,.2/.1/1,.1/2.1/2).

Feature Invocation Tone

Played when a feature is implemented.

The default is 350@-16;*(.1/.1/1).

Ring1 Cadence

Cadence script for distinctive ring 1.

The default is 60(2/4).

Ring2 Cadence

Cadence script for distinctive ring 2.

The default is 60(.8/.4,.8/4).

Ring3 Cadence

Cadence script for distinctive ring 3.

The default is 60(.4/.2,.4/.2,.8/4).

Ring4 Cadence

Cadence script for distinctive ring 4.

The default is 60(.3/.2,1/.2,.3/4).

Ring5 Cadence

Cadence script for distinctive ring 5.

The default is 1(.5/.5).

Ring6 Cadence

Cadence script for distinctive ring 6.

The default is 60(.2/.4,.2/.4,.2/4).

Ring7 Cadence

Cadence script for distinctive ring 7.

The default is 60(.4/.2,.4/.2,.4/4).

Ring8 Cadence

Cadence script for distinctive ring 8.

The default is 60(0.25/9.75).

CWT1 Cadence

Cadence script for distinctive CWT 1.

The default is 30(.3/9.7).

CWT2 Cadence

Cadence script for distinctive CWT 2.

The default is 30(.1/.1,.1/9.7).

CWT3 Cadence

Cadence script for distinctive CWT 3.

The default is 30(.1/.1,.1/.1,.1/9.7).

CWT4 Cadence

Cadence script for distinctive CWT 4.

The default is 30(.1/.1,.3/.1,.1/9.3).

CWT5 Cadence

Cadence script for distinctive CWT 5.

The default is 1(.5/.5).

CWT6 Cadence

Cadence script for distinctive CWT 6.

The default is 30(.1/.1,.3/.2,.3/9.1).

CWT7 Cadence

Cadence script for distinctive CWT 7.

The default is 30(.3/.1,.3/.1,.1/9.1).

CWT8 Cadence

Cadence script for distinctive CWT 8.

The default is 2.3(.3/2).

Ring1 Name

Name in an INVITE Alert-Info Header to pick distinctive ring/CWT 1 for the inbound call.

The default is Bellcore-r1.

Ring2 Name

Name in an INVITE Alert-Info Header to pick distinctive ring/CWT 2 for the inbound call.

The default is Bellcore-r2.

Ring3 Name

Name in an INVITE Alert-Info Header to pick distinctive ring/CWT 3 for the inbound call.

The default is Bellcore-r3.

Ring4 Name

Name in an INVITE Alert-Info Header to pick distinctive ring/CWT 4 for the inbound call.

The default is Bellcore-r4.

Ring5 Name

Name in an INVITE Alert-Info Header to pick distinctive ring/CWT 5 for the inbound call.

The default is Bellcore-r5.

Ring6 Name

Name in an INVITE Alert-Info Header to pick distinctive ring/CWT 6 for the inbound call.

The default is Bellcore-r6.

Ring7 Name

Name in an INVITE Alert-Info Header to pick distinctive ring/CWT 7 for the inbound call.

The default is Bellcore-r7.

Ring8 Name

Name in an INVITE Alert-Info Header to pick distinctive ring/CWT 8 for the inbound call.

The default is Bellcore-r8.

Ring Waveform

Waveform for the ringing signal. Choices are Sinusoid or Trapezoid. The default is Trapezoid.

Ring Frequency

Frequency of the ringing signal. Valid values are 10–100 (Hz). The default is 20.

Ring Voltage

Ringing voltage. Choices are 60–90 (V). The default is 85.

CWT Frequency

Frequency script of the call waiting tone. All distinctive CWTs are based on this tone.

The default is 440@-10.

Hook Flash Timer Min

Minimum on-hook time before off-hook qualifies as hook-flash. For values, less than this, the on-hook event is ignored. Range: 0.1–0.4 seconds.

The default is 0.1.

Hook Flash Timer Max

Maximum on-hook time before off-hook qualifies as hook-flash. For values greater than this, the on-hook event is treated as on-hook (no hook-flash event). Range: 0.4–1.6 seconds.

The default is 0.9.

Callee On Hook Delay

Phone must be on-hook for at least this length of time in sec before the WRP500 tears down the current inbound call. This does not apply to outbound calls. Range: 0–255 seconds.

The default is 0.

Reorder Delay

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

The default is 5.

Call Back Expires

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

The default is 1800.

Call Back Retry Intvl

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

The default is 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 WRP500 still considers the call as failed and keeps on retrying.

The default is 0.5.

VMWI Refresh Intvl

Interval between VMWI refresh to the CPE.

The default is 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.

The default is 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.

The default is 3.

CPC Delay

Delay in seconds after caller hangs up when the WRP500 starts removing the tip-and-ring voltage to the attached equipment of the called party. Range: 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.

The default is 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 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.

The default is 0 (CPC disabled).

Call Return Code

This code calls the last caller.

The default is *69.

Call Redial Code

Redials the last number called.

The default is *07.

Blind Transfer Code

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

The default is *98.

Call Back Act Code

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

The default is *66.

Call Back Deact Code

Cancels a callback.

The default is *86.

Call Back Busy Act Code

Starts a callback when the last outbound call is busy.

The default is *05

Cfwd All Act Code

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

The default is *72.

Cfwd All Deact Code

Cancels call forwarding of all calls.

The default is *73.

Cfwd Busy Act Code

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

The default is *90.

Cfwd Busy Deact Code

Cancels call forwarding of busy calls.

The default is *91.

Cfwd No Ans Act Code

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

The default is *92.

Cfwd No Ans Deact Code

Cancels call forwarding of no-answer calls.

The default is *93.

Cfwd Last Act Code

Forwards the last inbound or outbound calls to the extension specified after the activation code.

The default is *63.

Cfwd Last Deact Code

Cancels call forwarding of the last inbound or outbound calls.

The default is *83.

Block Last Act Code

Blocks the last inbound call.

The default is *60.

Block Last Deact Code

Cancels blocking of the last inbound call.

The default is *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.

The default is *64.

Accept Last Deact Code

Cancels the code to accept the last outbound call.

The default is *84.

CW Act Code

Enables call waiting on all calls.

The default is *56.

CW Deact Code

Disables call waiting on all calls.

The default is *57.

CW Per Call Act Code

Enables call waiting for the next call.

The default is *71.

CW Per Call Deact Code

Disables call waiting for the next call.

The default is *70.

Block CID Act Code

Blocks caller ID on all outbound calls.

The default is *67.

Block CID Deact Code

Removes caller ID blocking on all outbound calls.

The default is *68.

Block CID Per Call Act Code

Blocks caller ID on the next outbound call.

The default is *81.

Block CID Per Call Deact Code

Removes caller ID blocking on the next inbound call.

The default is *82.

Block ANC Act Code

Blocks all anonymous calls.

The default is *77.

Block ANC Deact Code

Removes blocking of all anonymous calls.

The default is *87.

DND Act Code

Enables the do not disturb feature.

The default is *78.

DND Deact Code

Disables the do not disturb feature.

The default is *79.

CID Act Code

Enables caller ID generation.

The default is *65.

CID Deact Code

Disables caller ID generation.

The default is *85.

CWCID Act Code

Enables call waiting, caller ID generation.

The default is *25.

CWCID Deact Code

Disables call waiting, caller ID generation.

The default is *45.

Dist Ring Act Code

Enables the distinctive ringing feature.

The default is *26

Dist Ring Deact Code

Disables the distinctive ringing feature.

The default is *46.

Speed Dial Act Code

Assigns a speed dial number.

The default is *74.

Secure All Call Act Code

Makes all outbound calls secure.

The default is *16.

Secure No Call Act Code

Makes all outbound calls not secure.

The default is *17.

Secure One Call Act Code

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

The default is *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.)

The default is *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.

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.

Modem Line Toggle Code

Toggles the line to a modem.

The default is *99. Modem pass-through mode can be triggered only by pre-dialing this code.

FAX Line Toggle Code

Toggles the line to a fax machine.

The default is #99.

Referral Services Codes

These codes tell the WRP500 what to do when the user places the current call on hold and is listening to the second dial tone.

One or more *code can be configured into this parameter, such as *98, or *97|*98|*123, etc. Max total length is 79 chars. This parameter applies when the user places the current call on hold (by Hook Flash) and is listening to second dial tone. Each *code (and the following valid target number according to current dial plan) entered on the second dial-tone triggers the WRP500 to perform a blind transfer to a target number that is preceded by the service *code.

For example, after the user dials *98, the WRP500 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 WRP500 sends a blind REFER to the holding party with the Refer-To target equals to *98 target_number. This feature allows the WRP500 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 WRP500. You can empty the corresponding *code that you do not want the WRP500 to process.

Feature Dial Services Codes

These codes tell the WRP500 what to do when the user is listening to the first or second dial tone.

One or more *code can be configured into this parameter, such as *72, or *72|*74|*67|*82, etc. Max total length is 79 chars. This parameter applies when the user has a dial tone (first or second dial tone). Enter *code (and the following target number according to current dial plan) entered at the dial tone triggers the WRP500 to call the target number preceded by the *code. For example, after user dials *72, the WRP500 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 WRP500 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 WRP500. You can empty the corresponding *code that you do not want to the WRP500 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 w/o 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 WRP500 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 WRP500 send INVITE *73@..... as usual when user dials *73.

Prefer G711u Code

Makes this codec the preferred codec for the associated call.

The default is *017110.

Force G711u Code

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

The default is *027110.

Prefer G711a Code

Makes this codec the preferred codec for the associated call.

The default is *017111

Force G711a Code

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

The default is *027111.

Prefer G729a Code

Makes this codec the preferred codec for the associated call.

The default is *01729.

Force G729a Code

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

The default is *02729.

Set Local Date (mm/dd)

Sets the local date (mm stands for months and dd stands for days). The year is optional and uses two or four digits.

Set Local Time (HH/mm)

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

FXS Port Impedance

Sets the electrical impedance of the FXS port. Choices are 600, 900, 600+2.16uF, 900+2.16uF, 270+750||150nF, 220+850||120nF, 220+820||115nF, or 200+600||100nF.

The default is 600.

FXS Port Input Gain

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

The default is -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.

The default is -3.

DTMF Playback Level

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

The default is -7.3.

DTMF Playback Length

Local DTMF playback duration in milliseconds.

The default is .1.

DTMF Playback Twist

Local DTMF playback duration.

The default is 1.3.

Caller ID Method

The following choices are available:

• 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 CPE 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 CPE after DTAS for CIDCW. Polarity reversal is applied only if equipment is on hook.

The default is Bellcore(N.Amer, China).

Caller ID FSK Standard

The WRP500 supports bell 202 and v.23 standards for caller ID generation. Select the FSK standard you want to use, bell 202 or v.23.

The default is bell 202.

Feature Invocation Method

Select the method you want to use, Default or Sweden default. The default is Default.

Line Enable

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

The default is 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.

The default is 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 WRP500 ends this call with a SIP BYE message. The range is 0 to 255 seconds (0 means that the session refresh is disabled).

The default is 30.

SAS Inbound RTP Sink

This setting works around devices that do not play inbound RTP if the streaming audio server line declares itself as a send-only device and tells the client not to stream out audio. Enter a Fully Qualified Domain Name (FQDN) or IP address of an RTP sink; this value is used by the streaming audio server line in the SDP of its 200 response to an inbound INVITE message from a client.

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 a FQDN or IP address of a 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 and, if specified, in the m = line of the SDP. If this value is not specified or 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.

The default value is empty.

NAT Mapping Enable

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

The default is no.

NAT Keep Alive Enable

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

The default is 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.

The default is $ 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.

The default is $ PROXY.

SIP ToS/DiffServ Value

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

The default is 0x68.

SIP CoS Value [0-7]

CoS value for SIP messages.

The default is 3.

RTP ToS/DiffServ Value

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

The default is 0xb8.

RTP CoS Value [0-7]

CoS value for RTP data.

The default is 6.

Network Jitter Min/Max

Determines how jitter buffer range of WRP500 when Network Jitter Mode is adaptive. Jitter buffer size is adjusted dynamically.

The default value of Network Jitter Min is 10ms.

The default value of Network Jitter Max is 200ms.

Network Jitter Mode

Specify whether the jitter buffer should be adjusted or use some constant interval value. Select the appropriate setting: adaptive, static.

The default is adaptive.

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. The default is UDP.

SIP Port

Port number of the SIP message listening and transmission port.

The default is 5060.

SIP 100REL Enable

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

The default is no.

EXT SIP Port

The external SIP port number.

Auth Resync-Reboot

If this feature is enabled, the WRP500 authenticates the sender when it receives the NOTIFY resync reboot (RFC 2617) message. To use this feature, select yes. Otherwise, select no.

The default is yes.

SIP Proxy-Require

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

SIP Remote-Party-ID

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

The default is yes.

SIP GUID

The Global Unique ID is generated for each line for each device. When it is enabled, the WRP500 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. This feature was requested by Bell Canada (Nortel) to limit the registration of SIP accounts.

The default is no.

SIP Debug Option

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

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

The default is none.

RTP Log Intvl

The interval for the RTP log. The default value is 0.

Restrict Source IP

If Lines 1 and 2 use the same SIP Port value and the Restrict Source IP feature is enabled, the proxy IP address for Lines 1 and 2 is treated as an acceptable IP address for both lines. To enable the Restrict Source IP feature, select yes. Otherwise, select no. If configured, the WRP500 will drop all packets sent to its SIP Ports originated 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).

The default is no.

Referor Bye Delay

Controls when the WRP500 sends BYE to terminate stale call legs upon completion of call transfers. Multiple delay settings (Referor, Refer Target, Referee, and Refer-To Target) are configured on this screen. For the Referor Bye Delay, enter the appropriate period of time in seconds.

The default is 4.

Refer Target Bye Delay

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

The default is 0.

Referee Bye Delay

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

The default is 0.

Refer-To Target Contact

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

The default is no.

Sticky 183

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

The default is no.

Auth INVITE

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

Use Anonymous With RPID

Set value of Remote Party ID to “anonymous, yes”

Use Local Addr in FROM

Use IP address in From header, no

Reply 182 On Call Waiting

Send 182 response when enter call waiting, no

Blind Attn-Xfer Enable

Enables the WRP500 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 WRP500 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.

The default is no.

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.

The default is yes.

MoH server

Address of music on hold server

Conference Bridge URL

URL of Conference server

Proxy

SIP proxy server for all outbound requests.

Outbound Proxy

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

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.

The default is 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.

The default is yes.

Register

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

The default is 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.

The default is no.

Register Expires

Allow answering inbound calls without successful (dynamic) registration by the unit. If proxy responded to REGISTER with a smaller Expires value, the WRP500 will renew registration based on this smaller value instead of the configured value. If registration failed with an Expires too brief error response, the WRP500 will retry with the value given in the Min-Expires header in the error response.

The default is 3600.

Ans Call Without Reg

Expires value in sec in a REGISTER request. The WRP500 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

Use DNS SRV

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

The default is no.

DNS SRV Auto Prefix

If enabled, the WRP500 will automatically prefix the Proxy or Outbound Proxy name with _sip._udp when performing a DNS SRV lookup on that name.

The default is no.

Proxy Fallback Intvl

This parameter sets the delay (sec) after which the WRP500 will retry from the highest priority proxy (or outbound proxy) servers after it has failed over to a lower priority server. This parameter is useful only if the primary and backup proxy server list is provided to the WRP500 via DNS SRV record lookup on the server name. (Using multiple DNS A record per server name does not allow the notion of priority and so all hosts will be considered at the same priority and the WRP500 will not attempt to fall back after a fail over).

The default is 3600

Proxy Redundancy Method

The WRP500 will make an internal list of proxies returned in DNS SRV records. In normal mode, this list will contain proxies ranked by weight and priority.

if Based on SRV port is configured the WRP500 does normal first, and also inspect the port number based on 1st proxy’s port on the list.

The default is Normal.

Voice Mail Server

Enter the URL or IP address of the server.

Mailbox Subscribe Expires

Expiry time to the voice mail server. The time to send another subscribe message to the voice mail server. The default is 2147483647.

Display Name

Display name for caller ID.

User ID

Extension number for this line.

Password

Password for this line.

Use Auth ID

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

The default is no.

Auth ID

Authentication ID for SIP authentication.

Directory Number

Enter the number for this line.

Call Waiting Serv

Enable Call Waiting Service.

The default is yes.

Block CID Serv

Enable Block Caller ID Service.

The default is yes.

Block ANC Serv

Enable Block Anonymous Calls Service

The default is yes.

Dist Ring Serv

Enable Distinctive Ringing Service

The default is yes.

Cfwd All Serv

Enable Call Forward All Service

The default is yes.

Cfwd Busy Serv

Enable Call Forward Busy Service

The default is yes.

Cfwd No Ans Serv

Enable Call Forward No Answer Service

The default is yes.

Cfwd Sel Serv

Enable Call Forward Selective Service

The default is yes.

Cfwd Last Serv

Enable Forward Last Call Service

The default is yes.

Block Last Serv

Enable Block Last Call Service

The default is yes.

Accept Last Serv

Enable Accept Last Call Service

The default is yes.

DND Serv

Enable Do Not Disturb Service

The default is yes.

CID_Serv

Enable Caller ID Service

The default is yes.

CWCID Serv

Enable Call Waiting Caller ID Service

The default is yes.

Call Return Serv

Enable Call Return Service

The default is yes.

Call Redial Serv

Enable Call Redial Service.

Call Back Serv

Enable Call Back Service.

Three Way Call Serv

Enable Three Way Calling Service. Three Way Calling is required for Three Way Conference and Attended Transfer.

The default is yes.

Three Way Conf Serv

Enable Three Way Conference Service. Three Way Conference is required for Attended Transfer.

The default is yes.

Attn Transfer Serv

Enable Attended Call Transfer Service. Three Way Conference is required for Attended Transfer.

The default is yes.

Unattn Transfer Serv

Enable Unattended (Blind) Call Transfer Service.

The default is yes.

MWI Serv

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

The default is yes.

VMWI Serv

Enable VMWI Service (FSK).

The default is yes.

Speed Dial Serv

Enable Speed Dial Service.

The default is yes.

Secure Call Serv

Enable Secure Call Service.

The default is yes.

Referral Serv

Enable Referral Service. See the Referral Services Codes parameter for more details.

The default is yes.

Feature Dial Serv

Enable Feature Dial Service. See the Feature Dial Services Codes parameter for more details.

The default is yes.

Service Announcement Serv

Enable Service Announcement Service.

The default is no.

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.

The default is G711u.

Second Preferred Codec

Second 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: Unspecified, G711u, G711a, G729a.

The default is Unspecified.

Third Preferred Codec

Third 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: Unspecified, G711u, G711a, G729a.

The default is Unspecifie d.

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.

The default is no.

Silence Supp Enable

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

The default is no.

G729a Enable

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

The default is yes.

Echo Canc Enable

To enable the use of the echo canceler, select yes. Otherwise, select no.

The default is yes.

Echo Supp Enable

To enable the use of the echo suppressor, select yes. Otherwise, select no.

The default is yes.

FAX CED Detect Enable

To enable detection of the fax Caller-Entered Digits (CED) tone, select yes. Otherwise, select no.

The default is yes.

FAX V21 Detect Enable

To enable detection of the fax v.21 signal, select yes. Otherwise, select no.

The default is yes.

FAX Passthru Codec

Select the codec for fax passthrough, G711u or G711a.

The default is G711u.

DTMF Process INFO

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

The default is yes.

FAX Codec Symmetric

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

The default is yes.

FAX Passthru Method

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

The default is NSE.

DTMF Tx Method

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

The default is Auto.

FAX Process NSE

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

The default is yes.

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.

The default is None.

Release Unused Codec

This feature allows the release of codecs not used after codec negotiation on the first call, so that other codecs can be used for the second line. To use this feature, select yes. Otherwise, select no.

The default is yes.

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.

The default is 1.

FAX Tone Detect Mode

If you want the Gateway to detect the fax tone whether the Gateway is a caller or callee, select caller or callee. If you want the Gateway to detect the fax tone only if the Gateway is the caller, select caller only. If you want the Gateway to detect the fax tone only if the Gateway is the callee, select callee only.

This parameter has three possible values:

caller or callee - The WRP500 will detect FAX tone whether it is callee or caller

caller only - The WRP500 will detect FAX tone only if it is the caller

callee only - The WRP500 will detect FAX tone only if it is the callee

The default is caller or callee.

FAX Enable T38

Set to yes to enable fax T.38 mode

FAX T38 ECM Enable

Set to yes to enable T38 error correction mode

*xx

Allow arbitrary 2 digit star code

[3469]11

Allow x11 sequences

0

Operator

00

International Operator

[2-9]xxxxxx

US local number

1xxx[2-9]xxxxxx

US 1 + 10-digit long distance number

xxxxxxxxxxxx.

Everything else (International long distance, FWD,...)

Dial Plan

Dial plan script for this line.

The default is (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.)

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

Example 1:

Example 2:

Example 3:

Enable IP Dialing

Enable or disable IP dialing.

If IP dialing is enabled, one can dial [user-id@]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.

The default is no.

Emergency Number

Comma separated list of emergency number patterns. If outbound call matches one of the pattern, the WRP500 will disable hook flash event handling. The condition is restored to normal after the phone is on-hook. Blank signifies no emergency number. Maximum number length is 63 characters.

The default is blank.

Idle Polarity

Polarity before a call is connected: Forward or Reverse.

The default is Forward.

Caller Conn Polarity

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

The default is Forward.

Callee Conn Polarity

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

The default is Forward.

Cfwd All Dest

Forward number for Call Forward All Service

The default is blank.

Cfwd Busy Dest

Forward number for Call Forward Busy Service. Same as Cfwd All Dest.

The default is blank.

Cfwd No Ans Dest

Forward number for Call Forward No Answer Service. Same as Cfwd All Dest.

The default is blank.

Cfwd No Ans Delay

Delay in sec before Call Forward No Answer triggers. Same as Cfwd All Dest.

The default is 20.

Cfwd Sel1- 8 Caller

Caller number pattern to trigger Call Forward Selective 1, 2, 3, 4, 5, 6, 7, or 8.

The default is blank.

Cfwd Sel1 - 8 Dest

Forward number for Call Forward Selective 1, 2, 3, 4, 5, 6, 7, or 8.

Same as Cfwd All Dest.

The default is blank.

Block Last Caller

ID of caller blocked via the Block Last Caller service.

The default is blank.

Accept Last Caller

ID of caller accepted via the Accept Last Caller service.

The default is blank.

Cfwd Last Caller

The Caller number that is actively forwarded to Cfwd Last Dest by using the Call Forward Last activation code

The default is blank.

Cfwd Last Dest

Forward number for the Cfwd Last Caller parameter.

Same as Cfwd All Dest.

The default is blank.

Speed Dial 2-9

Target phone number (or URL) assigned to speed dial 2, 3, 4, 5, 6, 7, 8, or 9.

The default is blank.

CW Setting

Call Waiting on/off for all calls.

The default is yes.

Block CID Setting

Block Caller ID on/off for all calls.

The default is no.

Block ANC Setting

Block Anonymous Calls on or off.

The default is no.

DND Setting

DND on or off.

The default is no.

CID Setting

Caller ID Generation on or off.

The default is yes.

CWCID Setting

Call Waiting Caller ID Generation on or off.

The default is yes.

Dist Ring Setting

Distinctive Ring on or off.

The default is yes.

Secure Call Setting

If yes, all outbound calls are secure calls by default.

The default is no.

Message Waiting

This value is updated when there is voice mail notification received by the WRP500. The user can also manually modify it to clear or set the flag. Setting this value to yes can activate stutter tone and VMWI signal. This parameter is stored in long term memory and will survive after reboot or power cycle.

The default is no.

Accept Media Loopback Request

Controls how to handle incoming requests for loopback operation. Choices are: Never, Automatic, and Manual, where:

• never —never accepts loopback calls; reply 486 to the caller
• automatic —automatically accepts the call without ringing
• manual —rings the phone first, and the call must be picked up manually before loopback starts.

The default is Automatic.

Media Loopback Mode

The loopback mode to assume locally when making call to request media loopback. Choices are: Source and Mirror. Default is Source.

Note that if the WRP500 answers the call, the mode is determined by the caller.

Media Loopback Type

The loopback type to use when making call to request media loopback operation. Choices are Media and Packet. Default is Media.

Note that if the WRP500 answers the call, then the loopback type is determined by the caller (the WRP500 always picks the first loopback type in the offer if it contains multiple types.)

Ring1 - 8 Caller

Caller number pattern to play Distinctive Ring/CWT 1, 2, 3, 4, 5, 6, 7, 8.

The default is blank.

Default Ring

Default ringing pattern, 1 – 8, for all callers.

The default is 1.

Default CWT

Default CWT pattern, 1 – 8, for all callers.

The default is 1.

Hold Reminder Ring

Ring pattern for reminder of a holding call when the phone is on-hook.

The default is 8.

Call Back Ring

Ring pattern for call back notification.

The default is 7.