Notifications > GNS Utilities > GNS Configuration Tester Utility > Configuring VoIP Properties

Configuring VoIP Properties

The VoIP page of the GNS Configuration Tester defines parameters for the voice messaging keywords in the GNS configuration file and allows users to modify the SIP settings in the Gns.cfg file. The page also provides a test feature to connect and interact with the specified SIP server.

The following is an example of the VoIP page of the GNS Configuration Tester dialog box with its corresponding GNS configuration file settings:

GNS Configuration Tester VoIP page

GNS configuration file settings

VoIP page of GNS Configuration Tester with corresponding Keywords in the GNS Configuration File

The properties on the VoIP page of the GNS Configuration Tester dialog box are described here. These settings also can be changed using the Config File Manager.

Note: To configure your system to test VoIP in your specific environment, see your vendor-provided documentation.

To Configure the VoIP Properties in the GNS Configuration Tester Utility

  1. Start the GNS Configuration Tester and when it opens, click the VoIP tab. If this is the first time you have used this utility, the GNS Configuration File box will be empty.
  1. Type a valid file path to the .cfg file in the box, then click Load Config. If the file path does not contain an existing file, then the Load Config button will be disabled. The relevant parameters from the .cfg file will be loaded into the utility.

-OR-

  1. Click the folder icon to browse to the directory that contains the Gns.cfg file, select the file, and click Open. This will automatically load the relevant parameters from the .cfg file into the utility.
  1. In the Configuration area click the boxes next to the items you want to edit and perform the following, depending on the items you want to change.
  1. Type a Username and Password. Note that authentication is required when VOICE_PROTOCOL is set to "SIP".
  2. Type a Display name.
  3. Type the SIP server name and its Port number. Note that when VOICE_PROTOCOL is set to "SIP", these fields are required.
  4. Type the STUN server name and its Port number.
  5. Select the Protocol type from the drop down menu.
  1. In the Test area:
  1. Click the folder icon next to the Audio file field to select the audio file to be played by the SIP client when a call is placed or answered.
  2. Type the phone number to be called in the Phone # field.
  3. Click Connect to connect to the specified SIP server and its port.
  4. Click Disconnect to disconnect from the SIP server and port.
  5. Click Dial to call the number entered in the Phone # field.
  1. The Status box displays the date, time, and whether the action was successful.
  2. Click Close to exit the utility.

GNS Configuration Tester Properties — VoIP

Field Description Associated Gns.cfg File Keyword

GNS Configuration File

The path to the GNS configuration file (Gns.cfg). Click the folder icon to browse to the file on the CygNet host.

N/A

Configuration

Username

Specifies the user ID to register to the SIP host.

Note: When VOICE_PROTOCOL is set to "SIP", this keyword is required.

VOICE_SIP_HOST_USER

Password

Specifies the user password.

Note: When VOICE_PROTOCOL is set to "SIP", this keyword is required.

VOICE_SIP_HOST_PASSWORD

Display name

Specifies the display name that may be made available to recipients for identification (e.g. caller ID) purposes. Optional field.

VOICE_SIP_DISPLAY_NAME

SIP server

Specifies the name of the host server providing the SIP connection.

Note: When VOICE_PROTOCOL is set to "SIP", this keyword is required.

VOICE_SIP_HOST

Port

Specifies the port used by the SIP host for SIP connections.

Note: When VOICE_PROTOCOL is set to "SIP", this keyword is required.

VOICE_SIP_PORT

STUN server

Specifies the name of the host server providing STUN services for the SIP connection. Optional field.

VOICE_SIP_STUN_HOST

Port

Specifies the port used by the STUN host for STUN services. Optional field.

VOICE_SIP_STUN_PORT

Protocol

Specifies the SIP transport and media transfer protocols used by the SIP host for VoIP communications. Default is "UDP_RTP", which means that the SIP client will use UDP for SIP transport and RTP for media transfer.

Note: "UDP_RTP" is the only available value.

VOICE_SIP_PROTOCOL

Test

Audio file

The path to the WAV file played by the SIP client when a call is placed or answered using the GNS Configuration Tester utility.

Note: WAV files must be 8000Hz, 16000Hz, 32000Hz, or 48000Hz mono 16-bit PCM format.

N/A

Phone #

The number that is dialed when the Dial button is clicked. If the phone number of the VoIP account for the connected SIP server/user is called, the Answer button is enabled, allowing a user to answer the call within the utility.

N/A

Status

Displays the status of the connection.

N/A

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