User Guide
LateralAccessDevice
 

User Guide

Table of Contents

About This User Guide

Chapter 1: Introduction / Meet LAD

Chapter 2: Installation

Chapter 2-1: Installation Tips

Chapter 2-2: Logging Into LAD

Chapter 2-3: Connecting to the Internet

Chapter 3: The Main Menu

Chapter 4: Configuring a Wireless Network

Chapter 5: Connecting and Managing Clients

Chapter 5-1: Groups

Chapter 5-2: Ports

Chapter 6: SIP Server / VoIP PBX

Chapter 6-1: SIP Accounts

Chapter 6-2: Call Routing

Chapter 6-3: Call Queues

Chapter 6-4: Voicemail Boxes / VM Menus

Chapter 6-5: Conference Rooms

Chapter 6-6: SIP Switches

Chapter 6-7: Call Logs

Chapter 6-8: Audio Files

Chapter 6-9: Call Routing & Processing Examples

Chapter 7: Access Controls / Parental Controls

Chapter 8: Scheduling

Chapter 9: Ping Monitors

Chapter 10: Network Security / LateralFirewall

Chapter 11: DNS Server and LateralDNS

Chapter 12: Reports

Chapter 13: Ethernet Packet Capture

Chapter 14: Wireless Packet Capture, Monitoring and Reports

Chapter 15: Settings, NAT Forwarding / Port Mapping and Sys Info

Chapter 16: LPN Membership

Chapter 17: Troubleshooting

Call Routing & Processing Examples

In these examples it is assumed that you have already programmed and registered your SIP devices with LAD and your client account(s) with your VoIP carrier(s). For instructions on programming client and server SIP accounts, see SIP Accounts.

How to Have Incoming Calls Go To VM Menu with Touchtone Options

To have your incoming calls go to a touchtone menu with a greeting and numerical options requires a client SIP account (for LAD to receive calls from your carrier), VM slot and VM menu. How Tos for setting up the Ring Groups and Call Queues that may be used as the targets for the VM menu's numerical options are covered further down on this page.

If you have not already done so, first set up the client SIP account to connect to your VoIP carrier. You will need the auth name and password from your VoIP carrier in order to set up the client SIP account.

Setting Up the SIP Account to Connect to Your Carrier

  1. From the Main Menu click on VoIP.
  2. Click on "Add SIP Account." This will open a new page.
  3. Give the SIP account a name for reference, perhaps using your carrier's name to nickname it and easily identify it later (the name has no bearing on functionality).
  4. Enter the credentials supplied by your VoIP carrier, putting the auth (aka auth name or authorization) in the Auth field and the password in the Password field.
    • You must enter the credentials exactly as supplied by your VoIP carrier. If they do not match, LAD will not be able to register the SIP account with the carrier.
  5. Checkmark "Active," "In Use," "Client" and "INVITE Tel."
  6. Save changes.

Next, create the VM Slot

Creating the VM Slot

  1. From the Main Menu click on VoIP.
  2. Click on "Add New VM Slot." This will open a new page.
  3. Give the SIP account a name for reference, perhaps using something like Main-Greeting to nickname it and easily identify it later (the name has no bearing on functionality).
  4. Record or upload the audio file to use as a greeting
    • To upload a file, use the form available on the VM slot page: click on "Choose File," then navigate to and select your desired file and click on "Upload File." See "Audio Files" for supported file formats.
    • To record the audio through LAD, go to the Record Audio section at the bottom of the VM slot page and click on "Generate Recording Tel Number." Call the number generated from a SIP device connected to LAD and begin recording your desired greeting. When done, hang up.
  5. Select your greeting from the "Default Prompt" dropdown menu.
  6. Select the default action you would like to happen if a caller does not choose an option from the "Action" dropdown menu.
    • If you choose Transfer, you must also select the destination to which to transfer the call in the "To" dropdown menu.
    • If you choose Record, you must also checkmark "Beep" to play a beep before recording starts.
  7. Checkmark either "Loop Prompt" to have the greeting repeat until the call is terminated, if the caller makes no selection, or "Replay Twice" to have the greeting repeat twice before proceeding to the default action (see previous step).
  8. Checkmark "Active" and "In Use."
  9. Save changes.

Then, create the VM Menu and numerical options.

Configuring the Touchtone Menu

  1. From the Main Menu click on VoIP.
  2. Click on "Add VM Menu." This will open a new page.
  3. Give the VM Menu a name for reference, perhaps using something like Main-Menu to nickname it and easily identify it later (the name has no bearing on functionality).
  4. Checkmark "Active" and "In Use."
  5. Save changes.
  6. Scroll down to the "Custom Menu Options" section.
  7. Assign the custom menu option a name for reference (the name has no bearing on functionality).
  8. Enter the number that you would like to trigger the custom menu option in the "Input" field.
  9. Choose the action to be triggered from the "Action" dropdown menu, either "Record," "Transfer" or "End Call."
    • If you select "Record," also checkmark "Beep."
    • If you select "Transfer," select the desired destination from the "To/From" dropdown menu.
    • If you wish the call to transfer to an external telephone number, such as a cell phone:
      1. Select "Transfer."
      2. Checkmark "Set Tel To."
      3. Enter the destination telephone number in the "Param" field.
  10. Checkmark "Active."
  11. Save Changes.
  12. Repeat for up to nine additional touchtone elements. If you need to provide more than ten numerical options to callers, you may create additional VM slots with VM menus and link to it either by setting the default action of the Main Menu to transfer to a second touchtone menu or by assigning one of the custom menu options to transfer to a second (or third, or fourth) touchtone menu.

Lastly, connect it all together.

Putting the Touchtone Menu into Service

  1. Go to the client SIP account created at the beginning of this How-To (the SIP account that registers with your VoIP carrier).
  2. Select the Main Greeting VM slot from the Default Route dropdown menu.
  3. Save changes.

How to Set Up Extensions

To set up extensions requires creating a SIP Group with SIP Tel Numbers corresponding to each extension and linking the SIP Group to all SIP accounts that you wish to be able to dial the extensions. These instructions assume that you have already created the SIP accounts to register your SIP devices.

First, create the Call Group.

Creating the SIP Group

  1. From the Main Menu click on VoIP.
  2. Click on "Add SIP Group." This will open a new page.
  3. Give the group a name for reference, such as "Extensions" to nickname it and easily identify it later (the name has no bearing on functionality).
  4. Select "DST ROUTE" from Type dropdown menu.
  5. Checkmark "Active" and "In Use."
  6. Save changes.

Next, add extensions to the SIP Group.

Adding Extensions to the SIP Group

  1. Scroll down to the SIP Tel Numbers section.
  2. Click on "Add SIP Tel Number." This will open a new page.
  3. In the Number field enter the numerical string desired for the extension.
  4. Select the desired destination for the extension from the To dropdown menu.
  5. Checkmark "Exact Match," "Active" and "In Use."
  6. Save changes.
  7. Repeat for the next extension.

Lastly, add the Extensions SIP Group as one of the route groups for the applicable SIP accounts.

Linking the Extensions Group to SIP Accounts

  1. From the VoIP Menu open the SIP account(s) that you would like to be able to use extensions.
  2. Go to the Call Routing section and select the Extensions SIP Group from one of the Route Group dropdown menus.
  3. Save changes.
  4. Repeat for the next SIP account.

How to Set Up a Call Queue to Ring in a Staggered Sequence to Different Destinations

Setting up a call queue to ring a first ring group once a call enters the call queue, a second ring group if a call remains unanswered three minutes after entering the call queue and a voicemail box if a call remains unanswered four minutes after entering the call queue requires creating a call queue and populating it with audio and ring destinations and creating a call route to the call queue by creating a destination route SIP Group. This example assumes that the two destination ring groups and voicemail box have already been created.

First, create the call queue:

Creating a Call Queue

  1. From the VoIP Menu click on "Add Call Queue." This will open a new page.
  2. Give the call queue a name of your choosing, such as "CQ-CustomerService."
  3. Checkmark "Active" and "In Use."
  4. Save changes.
  5. In the upload audio files section, click on "Choose File" and navigate to the audio you would like to use for the call queue's on hold music.
  6. Click on "Upload Audio File."
  7. Select the audio file that you just uploaded from the "Main" dropdown menu.
  8. Repeat for "Chime" and "Prompt" audio.
  9. Save changes.

Then, add the call destinations to the call queue. This example assumes that you have already created two ring groups (RingGroup1 and RingGroup2) and a voicemail box (VM-CustomerService) to use a destinations for the call queue.

Adding Destinations to the Call Queue

  1. Under "Ring To" select "RingGroup1" from the first dropdown menu.
  2. Enter "0" in the corresponding start time field to the left of the first dropdown menu.
  3. Checkmark the first "Enable" box.
  4. Select "RingGroup2" from the second dropdown menu under "Ring To."
  5. Enter "180" in the corresponding start time field to the left of the second dropdown menu.
  6. Checkmark the second "Enable" box.
  7. Select "VM-CustomerService" from the third dropdown menu under "Ring To."
  8. Enter "240" in the corresponding start time field to the left of the third dropdown menu.
  9. Save changes.

Next, create the destination route SIP Group and link it to the Call Queue CQ-CustomerService:

Link Call Queue to Destination Route SIP Group

  1. From the VoIP Menu click on "Add SIP Group." This will open a new page.
  2. Give the SIP Group a name for reference, such as Group-CQCS.
  3. From the "Type" dropdown menu, select Dest Route.
  4. Checkmark "Active" and "In Use."
  5. From the "Call Queue" dropdown menu, select Call Queue CQ-CustomerService.
  6. Save changes.

Lastly, either link a VM menu option to the sip group Group-CQCS or, for calls received from the VoIP provider to connect directly to the call queue, add the sip group Group-CQCS as the default route for the corresponding client SIP account.

Testing Call Routes

To check how a number would be routed based on the settings in LAD, use the Check Destination Number feature, which may be reached from the main VoIP page.

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