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AUTH2UserResourceIdentifier{userKey=8a00a0c862eadf5e0163170affe7001b, userName='null'}
AUTH1UserResourceIdentifier{userKey=8a00a0c87bd7898a017be2475bca0007, userName='null'}
REV5UserResourceIdentifier{userKey=8a00a0c85f4199b1015f7ea6e836000d, userName='null'}
REV6UserResourceIdentifier{userKey=8a00a0c85f4199b1015f7ea6e836000d, userName='null'}
REV3UserResourceIdentifier{userKey=8a00a0c867fcd99f0168116da9270007, userName='null'}
REV1UserResourceIdentifier{userKey=8a00a0c85985e6110159fde36227000e, userName='null'}

Introduction

Use the following parameters to configure the SIP gateway with global configuration settings.  

Global Configuration

ParameterDescription

Enable-SIP-Gateway

Default is False. This field enables the FXO gateway feature.

SIP-Virtual-IP

Read-only field. This value is hardcoded to 169.254.1.30. The field represents the IP where SIPGW will listen for all SIP/IP requests.

Max-FXO-Port

This field is read-only, with a value of 2. This field is the maximum number of FXO ports. 

Add-Dial-Out-Prefix

The range is from 0 to 9. The value to prepend to the Caller-ID received from PSTN to create the FROM field when trunking a call from the PSTN to the IP network. 

Register-With-SIP-Server

Register the FXO/GW with the SIP server using Override SIP FROM as SIP username.

SIP-SERVER-IP

This field is read-only, with a value of 169.254.1.120.  This is the IP address for the SWE Edge SIP communication.

Override-FROM-User-Name

Enter a message to override the system default SIP message FROM field. The FXO/GW registers with this name.

Note: When the Override FROM field is specified with the SIP username, it replaces Dial Out Prefix and CallerID.

SIP-Authentication-Name

Enter a name and password in the SIP Authentication Name and Password fields only if the SIP FXO/Line port needs to be authenticated.

SIP-Authentication-Password

Enter the password to authenticate the SIP FXO/Line port against the SIP softswitch or outbound proxy. This field is only needed if SIP FXO/Line port must be authenticated.

Override-FROM-Display-Name

Enter a display name to override the default “anonymous” display name.

If an override display name is not configured, the system uses information received from the PSTN as part of the CallerID by default. If CallerID feature is not present on the PSTN line, the default is “anonymous.”

CO-Disconnect-Timer

The range is 0 - 60 seconds. Enter the time in seconds for the PSTN CO to terminate the connection.

This specifies the time after the SIP caller has hung up for the PSTN CO line to terminate the connection when the call originated from the PSTN through the FXO port.

The time is usually 12-13 seconds. During this defined time value, the FXO port is not be available for making any outbound calls to the PSTN.


Default is False. Select the checkbox to enable priority calling services for FXO ports. When priority call services are enabled, any call placed to a priority calling number from any FXS port or LAN-side SIP phone is routed on priority basis to the FXO port and connected.

Note: Enabling Priority Calling Services overrides (or hides) the following settings for individual ports:

• Enable InBound (from PSTN) two-stage dialing

• Forwarded To settings

Priority-Call-Number

This is the default callback extension. When priority calling is enabled, all incoming calls are forwarded to this number. When priority calling is not enabled, incoming calls are forwarded to the Forward-To number defined at Port-Level-Configuration.

Priority-Call-Window

Default is 0. Enter the amount of time in seconds during which an inbound call from PSTN is forwarded to the last IP caller that made an outbound priority call through that FXO port.

The window starts when an outbound priority call is completed. If the window has lapsed and an inbound call from the PSTN is made, the call is forwarded to the Default Callback Extension instead.

This field is only visible when Priority Calling Services are enabled. 

PSTN-Track-Window

Default is 0 seconds. The range is 0 - 20 seconds. Enter the amount of time that the gateway senses the port to check if the PSTN line is connected to the FXO port and the line is alive. If there is no analog signal during this sensing period, the call is rejected with a “486” error. 

Inter-Digit-Delay

Default is 300 ms. The range is 100 - 2000 ms. Enter the delay in milliseconds between dialed digits to the PSTN.