Can a DID Have a User ID?

Q: 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.

A: 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.

Can I Customize the Number of Rings?

Q: Is there a way to let the administrator and the user customize the number of rings (or amount of time) the system should wait before forwarding an incoming call into their voice-mail?

A: 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 Media Sever Extension + IVR Service + User's Extension + PIN or SIP Password

Can I Make Calls from a SIP ATA?

Q: Can I make calls from a SIP ATA into the Dial-Media IVR without having any "credential information" configured on the ATA?

A: You cannot make an ITSP (telephonic) call without authenticating yourself to the sip proxy. You are able to communicate with the IVR system, but you will not be able to make an ITSP call. You can however use the IVR and call a registered DID without charge. This is an IP to IP call, and you will find that many providers offer this as well, like Skype. Being able to call the IVR without authentication was a design decision, but again, unless the UA authenticates himself, he will not be able to make an ITSP (telephonic) call.

Can Users Enable/Disable Voice-mail?

Q: 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?

A: That is correct. Only the administrator can disable or enable the voice-mail service.

Can We Get a Copy of Dial-Talk?

Q: How can we get a copy of Dial-Talk?

A: The Dial-Talk softphone can be downloaded from the Dialexia Website at the following link:
http://www.dialexia.com/products/VoIP-solutions-SIP-softphone-free-download.html

First you must be a registered user of the site to download this software. You can register at this link.

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Can You Explain These DID Fields?

Q: 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.

A: 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.

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Can You Forward to an Int'l Number?

Q: We tried to implement call forwarding to an international destination, but were unsuccessful. Does the system support forwarding the call to an international number?

A: Yes, Dial-Gate supports call forwarding. To setup call forwarding, you will need to log on Dial-Gate with the web ID of the subscriber account you wish to configure. Once you log in, click on the settings tab, scroll to the section labeled “When my status is…” In the “This Number” field, type the destination number. You can forward the call to any registered user within the system, or forward the call to PSTN.

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Can't Assign Rate Tables to a Call Shop.

Q: I made a few rate tables to assign to one of my call shop accounts and when I try to assign them to the call shop it doesn't let me. Why is this?

A: If your call shop is owned by an agent account, you can only assign it a rate table that is owned by that  agent account. In contrast, you must assign the agent account a rate table which they do not own, otherwise they could edit their own rates.

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Do I Need a License for Registration?

Q: Do Dial-Gate® and its modules use a license when registering?

A: Yes, however when we issue the dongle, we take into consideration any of our services that require a  license and we add them to the number of licenses purchased.

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Do You Support Alphanumeric Filter Characters?

Q: Can Dial-Gate® support filters with alphanumeric characters?

A: Yes we can, however to add alphanumeric characters, use the pattern replacement menu in the filters section.

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Do You Support Gateway Fail-over?

Q: If I have two gateways with different priority values assigned, and the gateway with the higher priority fails, will Dial-Gate® continue send to that gateway?

A: Yes, Dial-Gate® will continue to terminate to the gateway with the highest priority level. See explanation below:
-Multiple Gateways with different priority values assigned.
Dial-Gate® will always try to terminate to the gateway that has the highest priority (we are assuming that both gateways terminate the same Dial prefixes). For example, suppose we have Gateway A with priority 100, and Gateway B with priority 90. Both gateways will terminate the prefix 1 +xxx+xxxxxxx. Gateway A’s network link goes down, but Gateway B continues to function.  Dial-Gate® receives an invite and process the call, he looks at his routing table and see two possible gateways. Despite the fact that Gateway A if offline, Dial-Gate® will still attempt to terminate to this gateway first (based on the gateways priority value). If no response if received within 32 sec, Dial-Gate® will attempt Gateway B (The default value is set to 32, this value can be configured between the ranges of 7-32 sec).
-Multiple Gateways with the same priority values assigned.
If Gateway A & B have the same priority, Dial-Gate® then implements the feature “Last recorded failure”, and will internally designate one gateway as primary the other as secondary.
Let’s use the example above:
Since Gateway A failed to respond, Dial-Gate® will record the time of the failure along with the IP of the gateway. Dial-Gate® then will internally recognize that Gateway B as the primary gateway. That being said, should Dial-Gate® receive a new request, he will terminate to Gateway B first despite the fact both gateways have the same priority levels.
Now suppose Gateway B fails to respond, then Dial-Gate® will attempt gateway A. If successful, Dial-Gate® will record the failure, and designate Gateway A as the primary gateway.

Does Dialexia Have a STUN Server?

A: Yes we do; the address is stun.dialexia.com using UDP port 3479.

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Explain My Negative And Positive Balances.

Q: Why when I add money to my subscriber accounts do they show their balance as a negative number and the prepaids show their balance as positive?

A: The way the balance is displayed for a subscriber is in the context of the system administrator instead of the account user, so money deposited in the account is considered a balance owed to that account, which if exceeded, will result in a positive balance owed to the owner, similar to a credit card invoice where the outstanding balance is positive and a credit to the cardholder would show as negative. Prepaid accounts only use a prepaid amount which has already been deposited and can not carry an unpaid balance so their value is always displayed simply as positive.

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Explain PAP2 Auto-Provisioning.

Q: When we auto-provision accounts on the PAP2, the line 2 of the PAP2 which was not assigned a number registers on the proxy as a guest. Doesn't that affect the performance of the server or CPE at the customer site if we have all these CPEs connected with the second line as guest? Can't we disable line 2 from using the auto-provisioning until we want to assign it another number?

A: When you create a profile for auto-provisioning, you have the option to leave the lines ‘empty’ or assigned to a ‘guest’. A default value of ‘Leave Empty’ is assigned when you first create the profile. You can changed these values at any time prior to or after you have created the profile.

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Explain the Inheritance Feature.

Q: How many times can I use the inheritance feature to create rate tables off of rate tables that are inheriting from other rate tables?

A: You can create up to three levels. For example, you create a child rate table called agent_cost, from the parent rate table called wholesale_cost. Afterwards, you decide to use inheritance once again, and create a second child rate table, called end_user_cost, from the agent_cost table.

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How Can I Authenticate Users?

Q: What is credential information? In other words, what are the possible methods that I can use to authenticate users?

A: 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.

How Can I Configure Voice-mail?

Q: I have noticed that when I set status to unavailable and forward to voice-mail, that this doesn't work. Maybe I have not set it up correctly?

A: In order for you to receive voice-mail, Dial-Media (the IVR server), must be configured properly. After I verified your system, I found that  Dial-Gate® was running with its default configuration. By default, voice-mail will not function unless you have performed the following setup:
-Dial-Media 413 is set as the default IVR.
-The IP Tone user accounts are set to receive voice messages (please see IP Tone configuration).
-Of the two points, you had only one correct, the other was never performed. I made the necessary changes to your system, and you will find that you can receive voice messages.
-There are two IVRs running, one for ITSP calls, and a second IVR for IP Tone services.
-To access the ITSP IVR dial 412, and to hear the IP Tone IVR, dial 413.
-To access your mailbox using an ATA or IP phone, dial 413 + 98 + ext + pin; or you can listen to your mail from the Web.

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How Can I Use Dial-Relay?

Q: How can I access the Dial-Relay settings page?

A: Navigate to http://yourproxyaddress:8080 and login with the following credentials:
User: admin
Password: admin

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How Can Rate Tables Change With Calls In Session?

Q: If I change a rate table on an agent or subscriber while calls are in session, which rate will be applied on the current calls?

A: All current calls will continue to use the rate table that was in affect at the time the invite was processed. All new calls will have the new rate table applied.

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How Can the System Play Credit & Air time?

Q: The system does not seem to play any message if a user doesn't have enough credit to make the call.

A: In order to the subscriber to hear his available credit and air time, "Play Credentials" must be selected in the User's account page.

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How Can We Make Calls Without Blocking CID?

Q: How we can offer the customers to make calls out with blocking their Caller ID? With my Linksys ATA, I have option *67 to enable CID clear, and *68 to enable it. I tried to make a call with a clear CID but the server rejected the call.

A: When Dial-Gate® receives an invite, it must challenge the user. The user then resends the invite with the proper authentication Dial-Gate® requires. By doing so, the user is providing the proxy his User ID. The system then checks if the user (in the from header) matches one of the user ID's aliases. In your case it does not, and that is why Dial-Manager sends a 406 (unacceptable) message.
Dial-Gate® does have a feature called "Block Caller ID" (found in Settings > Server Settings > IVR Services). To use these feature, the call must go through the IVR; for example:
- Media Server ID + IVR service + destination number
- 413 + 90 + 15144211151
The only drawback to this feature is that both UAs must use G711 (a Dial-Media limitation).

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How Can We Setup Extensions?

Q: How can we setup and dial extensions for customers.  For example, a customer with a DID 16466520149 wants to be able to dial another DID setup on our network with a DID 16466520150. How can they call each other by dialing only the last 4 digits of the number?

A: Whenever a SIP user wants to use a different number (alias/alternate number) associated to his DID, the admin can easily add any number to his account under Alternate Numbers. Simply open the Sip User’s Settings page, and scroll to the bottom where it says “Alternate Numbers”. Once there, click on add, and apply any number you wish to link to the SIP User’s account (any number used must be unique).

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How Can We Support Call Hunting?

Q: 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.

A: 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.

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How Do I Delete a DID User Account?

Q: What is the correct way to delete an IP Tone user account who's registered with a DID from my Numbers pool?

A: Option 1: Locate the DID number in the pool and double click on it. Change the status from assigned to suspend. Once this is done, the DID number will return back to the pool as suspended, and the IP Tone account will be automatically deleted.
Option 2: Open the account page of the subscriber who has the IP Tone account in question. Select the IP Tone account you want to delete. Click on the X symbol.

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How Do I Setup Multiple Logons?

Q: I need more than one person to access an agent or ITSP account through the Web Interface, how can I do this without forcing them to log each other off when they connect?

A: It is possible to make multiple Web users for any given account, as well as multiple administrative accounts. Separate Web users can view the same account without logging each other out, but may also overwrite changes made by the other Web user.

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How Do We Use the Realm Field?

Q: Talking about the DNS, you mentioned something about the "Realm" field the first day you installed the application. Could you please explain to us how we can benefit from this field, as I remember you said it helps in securing our server?

A: 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.

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I Get An Import Rate Destinations Error.

Q: When I import my rate destinations from a spreadsheet, I get a screen saying an “an error has occurred”.

A: This means that another program, other than The Dial-Gate® Web Interface itself, has returned an error. In this situation there are two likely causes. One of the columns may not be formatted as the correct type of data (should be number for prefix and value, not custom or number stored as text). If all your formatting is correct, check for a small cost value of “0.000x”. This number may be converted into scientific format in Excel's API. We accept values of up to 4 decimal places but recommend using only up to 3 whenever possible.

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I Have Trouble Importing from a Spreadsheet.

Q: When I try to import rates from a spreadsheet for an account owned by my agent, not all of the  destinations are added, and there is no error that occurs. I have ensured all proper formatting of my document to be imported. What am I doing wrong?

A: If a destination prefix is included for importation that is not in the E164 list, it will be skipped over. Check that all the destinations are included in the agent's rate table as well. If a destination is imported to an account owned by an agent it must also be a valid destination in the rate table applied to the agent.

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No Invoice Generated Error Message.

Q: When I create invoices, it says no invoice was generated, along with an error message.

A: As a general rule, we tell our clients to create a billing policy and run their invoices periodically. The  billing policy should take into consideration two factors: the type of client, and number of pending CDR records. That being said, most subscribers and IP Tone users can be billed on a monthly basis. (from the 1st to the end of a month). In fact this is especially true for IP Tone users that have a monthly service plan in affect. All other accounts (agents and ITSP's), can be billed more frequently due to the large amount of traffic each one globally generates (most clients adopt a bi-weekly billing cycle).

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Pending Vs. Posted CDRs.

Q: What is the difference between Pending and Posted CDRs?

A: A Pending CDR is a record for call activity which has not yet been invoiced. CDRs for calls that have been invoiced already will be shown as Posted CDRs.

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We've Changed the Server Settings.

Q: We've assigned new domain name for our server: sip1.ourcompany.com and changed the values in our server settings to the new DNS. Will the changes be applied immediately?

A: Whenever changes are made to the server settings, you are required to restart the services of the affected module. This is done using the Dial-Gate® Service Manager in the Windows System Tray.

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What Are The Maximum Exportable CDRs?

Q: What are the maximum numbers of CDR records I can export?

A: If you use HTML, the max is 12000 records; if exporting to Excel, the max is 8000 records.

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What is the Invite:Blocked Account Error?

Q: I have an ATA registered to one of the call shop booths and I tried to make a call. The sip logs say, “INVITE: BLOCKED_ACCOUNT”. Can you explain this error?

A: Normally this is due to the booth not being allowed to make a call. Please check to make sure the booth is activated.

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What is the PIN Number?

Q: What is the PIN number? I'm asking here from a technical side. During a SIP call, when does the  SIP server ask the ATA for the PIN and where is it configured in the ATA?

A: A PIN is a combination of numbers used to challenge a telephonic account. The SIP server asks the ATA for a PIN during one of two processes: A subscriber calls the media server (using a telephonic account, not an IP Tone account), the media server will ask to enter your PIN. An ATA is pre-configured with an account number and PIN (both of which exists on the SIP server). The ATA dials directly to the system, not using the IVR.

Which DTMF Values Do You Accept?

A: Dial-Gate® will only accept DTMF set to RFC 2833 (out-of-band).

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

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Which Router Ports Should Be Blocked?

Q: 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?

A: 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.

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Which VoIP Line Has Been Picked Up?

Q: Our customers who have two lines, one from the USA and another from the UK, and are having to dial 011 for Int'l calls from the USA number and 00 for Int'l calls from the UK, but since the lines are connected to a PBX, the user is unable to determine which VoIP line has been picked up from the available ones.

A: Currently, Dial-Gate filters out international prefixes (011 & 00), so long as your rate and route tables reflect country code + destination number as opposed to Int’l prefix + country code + destination number. In either case, Dial-Gate can handle both setups.

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Why Am I Missing Certain Features?

Q: Why doesn't my Dial-Gate® have the options for call shop accounts or assigning numbers to my subscribers and viewing registered users?

A: The number/DID assignment features and registered users are only available if you have requested  Dial-Gate® with IP Tone. The call shops are also not a standard feature and must be requested with the system to be used. If you perform an upgrade to your system and use a software license (license files), ensure that you back up the license first, if you do not see the IP Tone or Call Shop sections after upgrading you have most likely overwritten the license file with a demo version during installation.

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Why Can't I Hear the IVR?

A: In order to access the IVR, your IAD must have G711 uLaw configured as one of its available voice CODECs.

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Why Can't Users Retrieve Voice-mail?

Q: Why can't my users retrieve their voice-mail from their Web interface?

A: If your users have voice-mail but are unable to see their voice-mail details in The Web Interface, you are probably using unified voice-mail. With unified voice-mail, their messages are stored on the E-mail server and not the machine running Dial-Gate®, and their messages will not be listed on The Web Interface.

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Why Didn't the PAP2 Resynch?

Q: We made some changes in the device provisioning, and the PAP2 didn't resynch?

A: In the master.cfg file, the default resync value is set for every 12 hours. Therefore, the PAP2 will not request any updates until the 12 hours are up, or until the unit is restarted. You can modify the resync values for your company's needs.

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Why Do We Get a 484 Error Message?

Q: We are unable to access the features or IVR service from our accounts, if we dial 403, we get a 484 error message (address incomplete) and if we dial any of the IVR numbers we get the same error.

A: In order to use the IVR, make sure your UA has G711 enabled. To access the IVR, dial 413 or any DID that is associated to Dial-Media. For all available options, see IVR Services.

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Why Does My Logo Look Distorted?

Q: Why does my logo look distorted on my welcome page, it looks fine in the settings page?

A: The logo will be stretched to fit to the correct size if it is smaller than the recommended dimensions. For optimal display, use the best possible approximation to the recommended size of Width 160 x Height 60.

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Why is the PAP2 Resynch Hard-Coded?

Q: The resynch period for the PAP2 configuration in the .xml file has the number 43200 hard-coded which is 12 hours. Why is that hard-coded? Should the PAP2 resync period be a passed parameter?

A: The re-sync value was taken from the Linksys PAP2’s documentation. You can always change the default value by editing the master.cfg file of the PAP2 located in your SIP server.

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Why Isn't My Filter Being Applied?

Q: I have created a filter for the prefix “1514” (for example), but when I dial calls to that area the filter is not being applied. What is the problem?

A: If you are making a filter or permission, check that your expression ends with a % sign or a number of periods equal to the rest of the digits in the number dialed, if the expression does not consist of the whole destination number. Most other items require a prefix to create such as the routes or rates, but the filters and permissions will try to match an expression, which must be an exact match with the number dialed unless wildcards are used. Use the % sign to represent any string of numbers or a period to represent any single digit.

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