Notifications > Getting Started with VoIP

Getting Started with VoIP

After setting up a VoIP/SIP account through your enterprise’s IT department or VoIP experts, use the VoIP page of the GNS Configuration Tester (GnsConfig.exe) to test SIP settings and save them to your Gns.cfg file. Configuring the Gns.cfg file to use SIP as its VoIP protocol allows the GNS to send Voice (text-to-speech) and WAV notifications, and to call into the GNS to acknowledge notifications.

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.

For more information, see Configuring VoIP Properties.

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

GNS Configuration Tester VoIP page

GNS configuration file settings

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 Gns.cfg file in the box, then click Load Config. The relevant parameters from the .cfg file will be loaded into the utility. If the file path does not contain an existing file, then the Load Config button will be disabled.

-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 (provided by your enterprise’s IT department or VoIP experts). Note that authentication is required when VOICE_PROTOCOL is set to "SIP".
  2. Type a Display name if desired. This field is the name that will appear in the caller ID of the callee (may be server-dependent or carrier-dependent).
  3. Type the SIP server name (host name)and its Port number (provided by your enterprise’s IT department or VoIP experts). Note that when VOICE_PROTOCOL is set to "SIP", these fields are required.
  4. Type the STUN server name and its Port number. These options are only necessary if the client is a NAT client on a network behind a firewall, connecting to a SIP server outside of that network. Your enterprise’s IT department or VoIP experts should be able to provide this information if it is necessary.
  5. Select UDP/RTP as the Protocol type from the drop down menu.
  6. Click the Connect button. If the settings are correct, a message displays in the Status window confirming a successful connection and registration to the SIP server. If a different message is displayed, use the code to help troubleshoot the problem. If no error is shown in the window, and there is no connection confirmation, there may be a communication error. In this case, an error should appear after a short time.
  7. After a successful connection, proceed to the next section to select an Audio file to play.
  1. In the Test area:
  1. Click the folder icon next to the Audio file field to select an audio file to be played by the SIP client when a call is placed or answered.
  2. Note: The Audio file option only applies to the GNS Configuration Tester utility. It is not saved to the Gns.cfg file, and will not apply to calls placed to/from the GNS.

    Note: If the selected audio file sounds distorted or sped up when played over the call, select a PCM WAV file with a higher sample rate.

  1. Type the phone number to be called in the Phone # field.
  2. Note: The Phone # option only applies to GNS Configuration Tester utility. It is not saved to the Gns.cfg file, and will not apply to calls placed to/from the GNS.

  1. Click Connect to connect to the specified SIP server and its port.
  2. Click Dial to call the number entered in the Phone # field.
  3. When a call is established, test DTMF tones by pressing numbers on the keypad. The Status window confirms the date, time, DTMF tones received, and whether the action was successful.
  1. To test a call-in, dial the phone number of the VoIP account used to connect to the SIP server from a different phone (e.g., a cell phone or similar). Your enterprise’s IT department or VoIP experts should be able to provide this number. When the GNS Configuration Tester utility detects an incoming call, the Answer button will be enabled. Click Answer to answer the call. The selected Audio file will play.
  2. When a call-in is established, test DTMF tones by pressing numbers on the keypad. The Status window confirms any DTMF tones received.
  3. After confirming that testing was successful, click Save Config to write the options to the Gns.cfg file specified in the GNS Configuration File field.
  4. Click Close to exit the utility.

GNS Configuration Tester Properties — VoIP

The following table includes properties on the VoIP page of the GNS Configuration Tester dialog box and their associated Gns.cfg file keywords (when applicable). Values displayed as "configurable" in the Value used for VoIP column are provided by the user or by your enterprise’s IT department or VoIP experts.

Note: The Dialogic section of the Gns.cfg file is not used for VoIP and is not included in the following table.

GNS Configuration Tester Field Gns.cfg File Keyword Required for VoIP Value used for VoIP Description

GNS Configuration File

N/A

Yes

configurable

The path to the associated GNS configuration file (Gns.cfg).

 

Voice Messaging Section

     

N/A

VOICE_PROTOCOL

Yes

SIP

Voice messaging protocol.

N/A

RESPONSE_PHONE_NUMBER

No

configurable

Can be appended to the notification voice message so that the user can know what number to call to acknowledge the notification. Optional field.

Configuration section

SIP

     

N/A

VOICE_SIP_LINE_COUNT

Yes

configurable

The number of simultaneous call lines allowed for the SIP client.

Valid values:

1 to 2,147,483,647

Username

VOICE_SIP_HOST_USER

Yes

configurable

The user ID to register to the SIP host. Encrypted by utility.

Password

VOICE_SIP_HOST_PASSWORD

Yes

configurable

The user password. Encrypted by utility.

Display name

VOICE_SIP_DISPLAY_NAME

No

configurable

The display name to be made available to recipients for identification (e.g. caller ID) purposes. Optional field.

SIP server

VOICE_SIP_HOST

Yes

configurable

The name of the host server providing SIP connection.

Port

VOICE_SIP_PORT

Yes

configurable

The port used by the SIP host for SIP connections.

STUN server

VOICE_SIP_STUN_HOST

No

configurable

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

Port

VOICE_SIP_STUN_PORT

No

configurable

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

Protocol

VOICE_SIP_PROTOCOL

Yes

UDP_RTP

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.

Test Section

       

Audio file

N/A

No

configurable

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.

Phone #

N/A

No

configurable

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.

Status

N/A

 

N/A

Displays the status of the connection.

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