You can authenticate telephonic users using a PIN, or Caller ID. To authenticate IP Tone users, the registration is based on Auth ID and SIP password.

An IP Tone user is someone who can be reached from the PSTN world (provided he has a real DID assigned). A telephonic user can only call out, and this user will never receive a call.

By default the number of rings is set to three. This means that Dial-Manager will wait 18 seconds (6 seconds per ring), before it sends the call to voice-mail. There is no section in The Dial-Gate® Web Interface that allows a user to set the numbers of rings, however, we can cheat the system.

What you can do is forward to voice after xxx rings. Click on the radio button labeled “this number” and type 413 (this is the media server alias), then 99 (this is the code for voicemail), then the extension of the user followed by the domain.

For example:

 41399258220@domain.com

Is it true that users cannot enable/disable their voice-mail inbox? They can only change the E-mail notification option. To enable/disable voice-mail, must an administrator do it?

Yes, that is correct. Only the administrator can disable or enable voice-mail service.

 In Adding a new DID for a user, what is the role of the fields: "User ID" and "Number/DID"? From what I can see is that the Number/DID is the actual extension of the user, while the user id is a reference. In this way, the authentication information should not change in case we want to change the DID for the customer.

The User ID is the SIP authentication ID of the user. The DID, is basically the user's number or extension where he can be reached. By default, when you create an account, we take the DID number and add it to the User ID section. If you change the DID of a user, then you will need to adjust the ATA device to reflect the new DID; the same rule applies if you change the password. The User ID cannot be altered.

For example:

User Id: 7777

DID: 5145558888

Authenticate pwd: 7777

Lets say the user wants to change his DID, and you assign him 4169991111, now we have this:

User Id: 7777

DID: 4169991111

Authenticate pwd: 7777

As you can see, the User ID remains the same, as does the password.

When adding a new subscriber, in "IP Tone Telephony Configuration" there is no way to fill in the "User ID", the system just assigns the User ID to be the same as the DID. However, if the user already exists, and we want to add new DID for him, there will be an area to fill in the "User ID" information.

Answer:  This is correct; you cannot alter the User ID of an IP Tone user. This does not stop you from adding additional DID’s for a user. You can add additional DID’s as aliases. What this means is that a user will receive calls based on both DIDs, even though he may be registered with one DID in particular.

 How can we support call hunting? That is, assigning the same DID for more than one account. The idea here is to provide the same service as a PSTN phone provider, where you can combine as many as trunks as possible under the same DID.

Answer: Dial-Gate® currently does not support call hunting, nor can you assign the same DID to multiple accounts. You can however assign multiple aliases to one account, and have each user register using the same User ID, but with a different DID number.

SIP authentication is based on HTTP authentication. All usernames and passwords belong to a realm. When we get a challenge from a SIP server, we have to choose username and password based on the server's realm. The “realm” parameter is required for authentication when the proxy receives a request; Dial-Manager will use this realm as one of its authentication parameters.

SIP authentication is based on HTTP authentication. All usernames and passwords belong to a realm. When we get a challenge from a SIP server, we have to choose username and password based on the server's realm. The “realm” parameter is required for authentication when the proxy receives a request; Dial-Manager will use this realm as one of its authentication parameters.

SIP authentication is based on HTTP authentication. All usernames and passwords belong to a realm. When we get a challenge from a SIP server, we have to choose username and password based on the server's realm. The “realm” parameter is required for authentication when the proxy receives a request; Dial-Manager will use this realm as one of its authentication parameters.

In our last conference call, Dialexia recommended that we block some ports on our router to avoid attacks from the outside into our SQL Server. Could you please tell us what ports? Do you suggest any setup here? Can't we block these ports from inside the machine?

Answer : We strongly recommend that you block SQL ports 1433, and 1434. In addition to these ports, we also suggest that you block the port 5060 on all your gateways. Create an ACL list that will allow only your SIP server to communicate directly with your gateways. By doing so, we will achieve two goals:

1. This will certify that all SIP signalling will pass through the SIP proxy at all times, thus eliminating the possibility of the UA communicating directly with the gateway.

2. The added security that your gateways will not process illegal SIP requests from rogue SIP servers.

Where to implement these restrictions depends on how you have your network setup; you can implement it at the top of your network structure, for example on the first router in your network, or on your switch, or on your firewall.

If you have Windows 2003 running, then you can setup all of these restrictions at the server level using the Windows built-in firewall.

Note: DTMF is short for Dual Tone Multi-Frequency, the system used by touch-tone telephones. DTMF assigns a specific frequency (consisting of two separate tones) to each key so that it can easily be identified by a microprocessor.]

RFC 2833 is the Internet Society Network Working Group Request for Comments describing how to carry dual tone multi-frequency (DTMF) signaling, other tone signals and telephony events in RTP (Real Time Protocol) packets (Copyright 2000).