User Guide

SIP Accounts

LAD links up with VoIP carriers and VoIP clients (telephone sets, SIP apps and etc) using SIP accounts. To set up your SIP accounts requires the Auth and Password information for your account with your VoIP carrier and/or for your VoIP telephone set or other SIP device.

There are two methods for creating SIP accounts, one by creating SIP account pairs to link up LAD with both your carrier and your SIP device, which preserves the current calling configuration that you may have already set up with your VoIP carrier and shields your SIP devices from the Internet. The second option is to manually create individual SIP accounts for all of your devices and your VoIP carriers separately and then interlink them manually.

Unless you are using LAD to link accounts from your carrier directly to SIP devices (as described below when creating SIP Intercept Pair Accounts, in which case the calls would be directly passed through with no intervening routing), please see the next section on Call Routing after you have created your SIP accounts.

Creating SIP Account Pairs

Using this method to create paired client and server SIP accounts in LAD preserves the current calling functionalities that you already have set up with your VoIP carrier, as the paired client and server SIP accounts are linked within LAD, so that SIP traffic is passed through from the VoIP carrier through LAD to the VoIP telephone set or other SIP device and vice versa, but non-SIP related traffic would not be relayed. This increases security and privacy.

To create a linked pair of SIP accounts you will need the Auth and Password information from your carrier, which is the information your carrier provided to you for programming into your SIP devices:

1. From the Main Menu click on VoIP, then click on "Create SIP Intercept Pair Accounts." This will open a new page with a short form.
• If you exit this page without submitting any information, no SIP accounts will be created.
2. In the Name field enter a nickname of your choice for the SIP accounts. This name is for reference only and has no bearing on the SIP device's configuration.
3. In the Auth field enter the auth string.
4. In the Password field enter the password.
5. Click on "Submit" and the system will create and link a pair of SIP accounts populated with the Auth and Password information supplied. One account will attempt to register with your carrier, while the other will be available for your SIP device to register with.

You may find your newly created SIP accounts listed under "SIP Accounts" on the main VoIP page. The client account (i.e., the account LAD will use to register with your carrier) will take the name exactly as entered in the form, while the account with which your SIP device would register would begin with "Local" followed by the name as entered in the form.

Use the Auth and Password from the "Local" SIP account to program your VoIP phone or other SIP device / application.

Manually Configuring SIP Accounts

You may also manually configure individual SIP accounts. SIP accounts may act as either clients to your VoIP carrier or as servers to your SIP devices. Whether the SIP account is a client or server account is determined from LAD's perspective: if LAD would be acting as a client to a server, the SIP account should be marked as a client account. If LAD is acting as the server to a SIP device, do not mark it as a client account.

To create a client SIP account to register with your VoIP carrier:

1. From the Main Menu click on VoIP, then click on "Add SIP Account." This will open a new page.
• If you exit this page without saving any changes, no SIP account will be created.
2. In the Name field enter a nickname of your choice for the SIP accounts. This name is for reference only and has no bearing on the SIP device's configuration.
3. Enter the Auth and Password information from your VoIP carrier into their respective fields.
4. Checkmark "Active," "In Use," "Client" and "INVITE Tel."
5. Select a Default Route in the Call Routing Section for incoming calls on this account to ring to.
• Incoming calls on client accounts (i.e. calls sent to you from your VoIP carrier) may ring to VM Menus, Ring Groups, Conference Rooms, Call Queues and individual SIP Accounts.
6. Save changes.

You will know that SIP accounts have registered with your VoIP carrier when the "Expires" date shown on the SIP account is in the future, i.e., at a later date and time than the current date and time.

To create a server SIP account to which your SIP device will register:

1. From the Main Menu click on VoIP, then click on "Add SIP Account." This will open a new page.
• If you exit this page without saving any changes, no SIP account will be created.
2. In the Name field enter a nickname of your choice for the SIP accounts. This name is for reference only and has no bearing on the SIP device's configuration.
3. Enter the Auth and Password information, either what is currently used by your SIP device or that you will be using to program your SIP device.
• The Auth must conform to the format A@B, wherein A is a unique character string of your choice and B is a domain or IP address that is resolvable (e.g., "fredsphone@192.168.0.1").
• We recommend using LAD's default LAN IP address 192.168.0.1, however, if you have changed LAD's LAN IP address, you should use that IP address instead.
• You may also be able to use @my.lad, however, some telephone sets may not accept it as a valid domain.
4. Checkmark "Active," "In Use," "LAN Only," "Return 503 on failed reg" and "INVITE Tel."
5. Select a Default Route for the SIP account (see Call Routing for more information on routing calls).
6. Save changes.
7. Use the Auth and Password information from step 3 to program your VoIP telephone set (or other SIP device or application).

You will know that your SIP devices have registered with LAD when the "Expires" date shown on their SIP accounts are in the future, i.e., at a later date and time than the current date and time. Alternatively, check the telephone set for its indicator that it has registered.

You may find your newly created SIP accounts listed under "SIP Accounts" on the main VoIP page.

DTMF and Audio Codec Settings for Your VoIP Equipment

LAD supports the G.711a, G.711u and G.729a audio codecs, however, not all DTMF settings work equally well with all codecs (DTMF refers to the pickup of the tones corresponding to keys pressed during a telephone call, e.g, when pressing a number to make a selection from a touchtone menu.

• G.729a is a great codec for voice quality, however, it does not work well for InBand DTMF. If using the G.729a codec on your VoIP equipment, it is advisable to change the DTMF setting to INFO or any other option other than InBand.
• InBand DTMF works well with only G.711a and G.711u.

You may need to try different codecs and DTMF options to see which combination works best on your VoIP equipment.

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