Voip Switch Manual

VoipSwitch manual This page last changed on Feb 25, 2008 by admin.

VoipSwitch manual • • • • • • • • • • •

1.0 Main system 2.0 Clients 3.0 Destinations 4.0 Dialing plan 5.0 Tariffs 6.0 Browsing calls, reports, statistics, payments 7.0 Settings 8.0 Services 9.0 Invoices Changes Common UI elements

VoipSwitch system is built from main application VoipSwitch.exe and configuration application named VSM. VoipSwitch.exe must be working all the time when calls are going and VSM can be opened from button Config available on VoipSwitch for configuring all VoipSwitch function details. VSC is web version of VSM with similar functionality.

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Point 1.0 of manual describes VoipSwitch.exe itself and next points about VSM. Calls in Voipswitch are coming from Clients towards Gateways or GK/Registrars. So Clients send calls to Voipswitch while Gateways and GK/Registrars terminate them. Rules for sending calls are defined in Dialing Plan. To charge clients for calling or calculate cost of termination are used Tariffs . All calls must be authorized first as Clients from one of 6 available types. Calls, statistics and other reports are availalble in few parts of VSM. Invoices are self descriptive but Services are used to define automated tasks. Below are typical steps to start working with VoipSwitch: 1. Create the termination accounts. If you have to send calls to terminate GW in direct mode then you create an account in Gateways. Account in GK/Registrar to register VoipSwitch to gatekeeper or SIP Proxy. 2. Rules how to send calls depending on dialed numbers are defined in DialingPlan. 3. Tariffs must be created. Some tariffs can be used for clients, resellers and termination devices. All are defined in the same place available in VSM named Tariffs 4. Create clients to authorize calls coming to VoipSwitch. Wholesale clients can be add as GW clients with many ip numbers for one client. Clients calling from ip phones are added as Common clients. 5. All VoipSwitch working parameters can be adjusted in Services

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1.0 Main system This page last changed on Feb 23, 2008 by admin.

• Calls processing • ° Filter and display settings ° Reload settings and listeners actions • Logs window • Statistics • Registered Clients • ° Edit selected client • Gatekeepers • ° Synchronize with database ° Gatekeeper settings • Users

Calls processing This window is showing calls made by Clients (see Fig.1). You may customize the way calls are displayed with filters and maximum calls number (see Fig.2). All settings are described below.

All calls are shown in following manner: [ICON] Call to number: [DESTINATION NUMBER], [CALLER ID] ([CALL TYPE])

Icons are: New call is connecting. Call connected and active. Call not connected with some reason. Call finished properly. Fig.1 VoipSwitch calls processing window Call failed with some reason. [DESTINATION NUMBER] - this is number Client dials (send by gateway or Client's device) [CALLER ID] - this is Client's ID (sent by gateway or Client's device) [CALL TYPE] - short description of Client's connection, for example: • (H323 Reg) or (SIP Reg) - call from registered H323 or SIP device

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• (Callback call) - call initiated by VoipSwitch after client's call to callback trigger number • (H323) or (SIP) - call from H323 or SIP gateway

Filter and display settings Available commands in context menu (activated after right mouse click on calls window).

Fig.2 Calls processing window - context menu Context manu give you possibility to manipulate calls display settings, including: • Freeze call list - when you activate this option you can easily look through calls that was on the list - any new call will be shown • Maximim logs - this option will allow you to limit number of calls shown in the calls processing window. It is useful when you don't need to see all calls made but for example only last 200. In such case calls processing window is more readable and uses less system resources. • Filter - this option give you possibility to bound displayed calls. It is useful when you want to see only calls made by one Client or/and to relevant destination number (see Fig.3). When setting up filter only new calls are filtered. • Clear filter - resets current filter applied to default settings (to show all calls) • Clear list - this option removes all calls from list (it doesn't influence calls in database, just calls window is cleared)

Fig.3 Calls filter options In example on Fig.3 calls are filtered to show only callback Client calls or calls with destination number 48774560220. Every other calls will not be visible on the list, but of course this will not influence other calls.

Reload settings and listeners actions Context menu is shown on Fig. 2. Last two options from context menu allows you to reload settings and start or stop listeners.

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• Reload settings mean, that VoipSwitch will read and apply all changed settings). • Start listeners may be started and stopped. By default all listeners are started after VoipSwitch start. If you want to stop listeners (ie. when changing VoipSwitch version) just right mouse click on Calls window and choose "Stop listeners". When listenres are not running appriopriate information is shown on Calls window title bar (see Fig. 4).

Fig.4 VoipSwitch with stopped listeners. When listeners are not running new calls will not be connected.

Logs window This window is used to display startup parameters and informations about abnormal Clients operations (ie. calls limit reaching, unknown gateways call attempts).

Fig.5 VoipSwitch log window.

Statistics Statistics window is displaying in real time informations about current and past calls (since VoipSwitch start). There are four main sections - summary statistics, incoming calls, outgoing calls and Clients/Users counters. Below you can see exemplary Statistics window (Fig. 6) and short description of computed values:

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• Connections connected - sum of all successful connected calls since last VoipSwitch start • Total connections - sum of all calls since last VoipSwitch start (connected and failed) • ASR - (Answer seizure ratio) - is computed as: Connections connected / Total connections

• Incoming and outgoing calls: ° pending - currently pending calls (callls not yet connected) ° connected - current active calls ° total calls - total incoming/outgoing calls that has reached VoipSwitch since last start ° total connected calls - total successfuly connected incoming/outgoing calls that has reached VoipSwitch since last start ° H323 calls - all calls that was using H323 protocol Fig.6 VoipSwitch Statistics window.

° SIP calls - all calls that was using SIP protocol

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° ASR - answer seizure ratio for incoming/outgoing calls ° ACD - (Average incoming/outgoing call duration) Registered users - All currently registered users (more details in Registered clients section) Registered user calling - Sum of calls in progress made by registered users. Total users - sum of logged users via Portal/Web, Callshop, Callback module (described later) Users logged - currently logged users.

Registered Clients

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All clients are represended by icon, username and their IP address (public / private) Clients icons are: Registered SIP Client. Registered SIP Client with active call. Registered H323 Client.

Fig.7 VoipSwitch registered clients window.

Registered H323 Client with active call. Registered PC2Phone Client. Registered PC2Phone Client with active call.

Edit selected client There is possibility to edit registered Client settings without searching one in VSM or VSC. To do so just click on Client (right mouse button) and select Edit as shown on Fig. 8, VSM window with Client's detailed configuration will shown.

Fig.8 VoipSwitch registered clients edit dialog. After right click on Client some basic informations are also displayed (tariff, founds, prefixes and codecs).

Gatekeepers Gatekeepers window is displaying current state of all active gatekeepers. VoipSwitch has to be registered to gatekeeper in order to send a call there. Every gatekeeper login state is shown on Gatekeepers window (Fig. 9).

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After some changes in gatekeepers configuration you should reload settings. This may be done by right click on Gatekeepers window and choose (only one available) option named: Synchronize with database (Fig. 9). VoipSwitch will read and apply all gatekeepers settings.

Fig.9 VoipSwitch Gatekeepers reload settings.

Gatekeeper settings Gatekeeper settings shown after right click on one of listed gatekeepers (Fig. 10). You may see some simple statistics (calculated since VoipSwitch start) and Gatekeeper IP, name, H323 ID, E164 and codecs options.

Each gatekeeper has icon next to it's name describing current register status. There are only two possible icons as shown below: H323 Gatekeeper online (registered) SIP Registrar online (registered) Gatekeeper/Registrar offline (unregistered)

Fig.10 VoipSwitch Gatekeepers window with settings. There is also (Fig. 10) log information about gatekeeper login state and 3 buttons to Login, Logout and Reload data for gatekeeper.

Users This window is showing currently logged and past login/logout actions for Clients who use Web or Portal module (Web Dialer Clients) or standalone Callback, Callshop modules (Fig.11).

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For each Client login and logout time are displayed. Clients may have different icons with their names, it depends on their login state as shown below. Client online (connected) Fig.11 VoipSwitch Users window.

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Client offline (disconnected)

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2.0 Clients This page last changed on Feb 24, 2008 by admin.

• Introduction • Common features of clients • ° Login and password ° Tariff ° Currency ° Account state ° Prefixes ° Active state ° Personal data ° Reseller • GW clients • PC2Phone clients • GK clients • Callback Clients • IVR clients • Common clients • Callshop clients • Guest account • Automatic clients generation • ° Lot's propperties: ° Logins and passwords • Currencies management • ° Description ° Currency definition ° Adding ratio values ° Advanced - currency proccessing • Import-export clients

Introduction Every call coming to VoipSwitch must be authorized before processing. Voipswitch authorize calls from 6 types of clients that differ by functions, method of autorization and available options. Some features are the same for all kinds of clients.

Clients are added using VSM or VSC or by reseller through VSR pages. In addition automatic registration realized through Web or Portal is used to add clients. Type of clients available in VoipSwitch system 1. GW clients

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2. 3. 4. 5. 6. 7. 8.

PC2Phone clients GK clients Callback Clients IVR clients Common clients Callshop clients Guest account

Common features of clients Login and password It is used differently by every type of client. For GW clients it can be used to authorize every call. For GK , PC2Phone or Callshop clients it is used to log to the system. IVR clients use just a password as PIN number to authorize callers to use the IVR. One common functionality for all types of clients is loging to a web page using login and password. Every type of client has different information available there and can use it to get access to his account.

Tariff Tariff assigned to a client is used to: • calculate cost of a call for the client • estimate maximum time of connection • calculate the remaining time announced for IVR clients • limit available directions. If there is no matching prefix in tariff, the call will not be realized. The cost of every call is calculated using tariff right after disconnection. When tariff for a client changes in the future, all calls made untill this change won't be changed. The system will use new tariff only for new calls and old ones will be left unchanged. Details on how to define tariffs and how to use them in cost calculation are described here

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Currency This option allows assigning currency to a client so he or she can be charged in different currency that VoipSwitch owner is charges.

Details about currency support are described here

Account state Client must have some funds in the account to be able to make calls through Voipswitch. One exception is when tariff assigned to a client has 0 cost rates, but this is rather unusual. In most cases every call is charged and this amount is subtracted from client's account state value. When value reaches 0 the client will be blocked. Account state value can be modified only by adding payments. Using payment in comparison to direct modification account state value has one big advantage. Every change is recorded with date and optional description. There are 4 types of payments: 1. Prepaid - should be used after client is paid money. 2. Return - when it is necessary to return money to client this payment type should be used. Return payment cannot be higher than funds available on clients account. 3. Credit - adding fund with this payment type allows client Credit Balance to go below 0 and continue making calls. Total available credit for client is a summary of all credit payments made for him. It is not clear for some clients but we decided to build it this way to avoid problems with clients overusing accounts. If client really wants to have unlimited credit then it is possible to add big amount as credit payment. 4. Return credit - this payment decreases available credit for client.

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The most typical way to increase account state (balance) is to add payment. It can be done by VoipSwitch owner using VSM or VSC or by reseller. Reseller can add funds only to clients belonging to him. Clients can see history of payment on the web and recharge accounts in several ways. Methods of recharging are described here.

Prefixes This is a general name used for manipulating information being sent in a client's call. It is specified as • Dialing plan prefix • Tariff prefix • Caller id prefix First it must be explained how VoipSwitch processes calls coming from a client. After client authorization, VoipSwitch checks the dialed number. It must match the entries defined in Dialing Plan and in Tariff. Before searching the dialed number in dialing plan it can be modified by Dialing plan prefix. It will not change number used to find prefix in Tariff. To modify number before searching in tariff tariff prefix must be used. Caller id prefix is used to modify caller ID being sent to VoipSwitch from a client. Dialing plan prefix and tariff prefix modyfy the called number seperately for every given client. A rule defined in one place is not used for another. Every prefix is built from digits or characters. Modifcation of them is described in special section available here

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There are additional prefixes available for callback calls: • Source dialing plan prefix • Source tariff prefix • Source caller id Every callback call consists of two legs, which means that different rules can be set for modyfing number or caller id for every leg.

Active state Client can be active or not active. Not active client is forbidden to make any call but can still log on the web.

Personal data Every client has an option to write extended information about himself. Available fields are presented on figure below.

These information is used when creating invoices or sending warning emails defined in Services.

Reseller Client created in VoipSwitch can belong to reseller or he can be unassigned. Information about assigned reseller is presented in client definition and can be changed. However it is not recommended to do it manually. It is more secure to do it through the resellers pages.

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GW clients Those clients are used mostly for carriers and wholesale services. Other popular application is to authorize DID numbers being used to: • activating callback • calling to IVR scenarios • calling to devices and make charging them for answering Options available for GW client Login field is the username for this account. Password is the allocated password. These 2 fields are used to access the web page to see the CDR's. Also the Login@Password combination is used to match against the H323ID sent by the client in case that Authorise by login/password feature is enabled. For SIP clients login and password can also be used without adding client's ip.

DID source - allows to charge clients answering calls. It is useful with DID services when client is paying a monthly fee for the number and then additionally for every call answered using this number. This option will work with calls ending to PC2Phone, GK or Common clients. When checked, every such client will be charged for answering a call. Tariff assigned to this client will be used to calculate cost of a call. If a client doesn't have enough money to pay even one billing step, the call will not be connected. PIN source option is used for calls made to IVR system. Calling to PIN asking scenario allows to work with calling cards services. Only with this option checked GW client can connect to scenario with PIN name. Such call will be billed in two ways. SIM Source Supported codecs

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Allows the selection of 4 codecs groups depending on what client device can support. One codec has to be set as primary and it will be the default codec. Voipswitch supports group of codecs, meaning that if you select g723.1, all kind of g723.1 codecs will be allowed, including g723r63 and g723r53. Same thing for other codec groups. After selection of the codecs you can enable Use client codec to let VoipSwitch negotiate the right codec from the list with client device. Of course client's device has to be able to autonegotiate codecs. Please note that VoipSwitch acts differently in "proxy all" mode and in "proxy only signaling" mode. In "proxy all" VoipSwitch does not allow codec negotiation directly between endpoints and instead will negotiate itself with each endpoint in part. While in "proxy only signaling" the endpoints can negotiate directly the codecs, it is possible to choose any codec that both endpoints support, even those that are not listed in VoipSwitch settings. IP numbers are the list with authorized IP addresses. Cost of calls coming from ip assigned to a client is taken from his account. You can set here an unlimited number of addresses, but an IP can be enabled only for one GWClient at a time. Under the IP numbers list there is a field where to write the new addresses to be added in the list. Use the Add IP button after you fill it. To remove an IP from the list select it first and then click Remove IP. It is possible to add ip addresses in range. After clicking Add Range button the dialog as on screen below will appear.

There it should be set starting ip and ending ip. VoipSwitch will use them as boundaries to create

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appropriate entries in ip numbers list. For starting ip will be added 1 till it reaches ending ip. Connect client immediately Enable this only when all calls of a client do not connect to any destination. This will open the media channel immediately after routing but in most cases will generate also false billing because the calls will be declared answered immediately. So this feature is for extreme cases only. Do not use it for normal users. Calls limit Used to a limited number of concurrent calls being send from gateway. When number of calls is equal to this limit any new calls from this client will be rejected. This is also checked for calls in progress and connected apiece.

PC2Phone clients This type of client is for pc2phone dialer and web2phone page access only. Pc2phone is a proprietary application that allows clients who have a valid pc2phone client account to connect to the VoipSwitch and initiate and also to receive calls. This dialer uses particular communication ports and is not compatible with other systems. The settings for pc2phone clients are very simple and the fields have the same meanings as for GWclients. Pc2phone application always uses g723.1 codec group so there is no need for codec settings. PC2phone client is allowed to make only one call at the same time. This type of client is hot billed . Login and password defined for every client are used to log using pc2phone application

More about PC2Phone application is described here. Setting termination on PC2Phone client is described in dialing plan section of manual.

GK clients This client type is used for those devices behind NAT, or those that change the IP often or simply want to register with a user and pass only. The client will have to configure his device to register to VoipSwitch's

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Gatekeeper (when using h323 protocol) or Registrar (for SIP protocol) using the user and password he received. Also he will need to enter in his device configuration the IP of Voipswitch and the Gatekeeper name that is by default Gatekeeper (in case he uses h323). Parameters supported codecs, calls limit have the same meaning as desribed for GW clients. Login and password are used to log from device to VoipSwitch acting as Registrar or Gatekeeper. VoipSwitch will recognize automatically protocol being used to log. Clients of this type are hot billed but only when one port of device uses one login and password. Otherwise hot billing the function may not work, for example when one call is started and then second port using the same account calls. The system will not be able to calculate properly remaining account state, and account balance can go below zero. To eliminate this possibility calls limit value should be set to 1. After sucessfull loging to VoipSwitch a device will appear in Registered clients. Different icon will be used for h323 and sip devices. GK clients working with SIP protocol can be used also with VoipTunnel module. How to work with VoipTunnel and GK clients is described here

Callback Clients These clients can use different types of callback. Detailed description can be found in manual for callback system. Every connection made by callback client consist of 2 calls and both are charged. Main callback features are listed below: • After being connected to destination number a client can finish call and pick another number without disconnecting source leg of connection. • After setting appropriate scenario a client can hear account state and remaining time announcement after every call made. • There is an option available to charge source leg only if destinations were connected. • Separate dialing plan ,tariff or caller id prefix can be set for source and destination number used. Thanks to this it is possible to use different rates defined in the same tariff for source and destination numbers. Also different gateways can be used.

IVR clients Used in calling card service. Using this type of clients is possible only with VoipSwitch with IVR module. Every client must go to VoipSwitch through connection already authorized. Clients are using regular phones to call to special number redirected to VoipSwitch. Call from such gateway being connected to PSTN network and VoipSwitch is authorized as GW client with PIN source option checked. Then a client connects to scenario which asks about PIN number. After finishing one connection the user can pick another number without dialing access number again. Client can call to recharge scenario to add funds using special recharge codes or check account state on his account. Calling from the authorized phones without entering pin. Authorized phones can be added: • on the web, • by sending sms message

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• by calling to special scenario which will register phone number after successful login by pin

Common clients Special type of client which can be used to call from different services. Client of this type can log from • • • •

PC2Phone dialer, from ip phone, can call to IVR system and log using pin activate call using callback. VoipSwitch allows to use only one type of services at the same time. It is recommended to use this type of client for new clients. No matter what service is used to log to VoipSwitch ( from dialer or ip phone ) a client can receive calls. Calls redirection can be set in the same way as for other types of clients. It can be set one account for all services and no matter what service is used the same account is charged for calling.

Callshop clients This type of clients differs greatly from the others. Client of such type is used to log to VoipSwitch from callshop application. Callshop application is available as part of a callshop module. Every callshop definition consists of a number of cabins assigned. As a cabin can be used client of type Pc2Phone, GK registrar or GW.

Client assigned to callshop is used differently then unassigned. When callshop client ( to which specified cabin belongs ) is logged in VoipSwitch then the cost of every call is taken from callshop account, not from cabin account. Cabin account state should be set with 0 amount to avoid calling from it when callshop application is logged off.

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When callshop client account will reach 0 then any cabin will be blocked from calling. Tariff assigned to cabin is separate from tariff assigned to callshop. This tariff is used to calculate end user prices and is higher then callshop tariff. Difference between those tariffs is profit for callshop. Callshop client can change cabins rates through the web interface so such client has a right to set rates charged from clients. It is important that tariff used for cabins be different for every callshop client because if it will be the same changing it causes changes for other cabins assigned to other callshops. More about callshop application is defined in Callshop manual.

Guest account Special feature allowing to call from unauthorized devices. It can be turned on using VSM->Settings->VoipSwitch. There is a combox box with list of GW clients. If any client is chosen from it then calls not authorized ( normally rejected ) will be accepted and assigned to this client.

Automatic clients generation For every type of client it is possible to generate clients automatically in lots. All clients are assigned to a lot identified by name. Later it can be easily managed to change tariff, modify account state for all clients, export , activate or deactivate or delete. Generated lot can be assigned to reseller. Automatic client generation is available in Clients node of VSM or VSC application.

After clicking it the list of lots will appear

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Every row in this list is describing one lot. There is name of a lot, number of clients and type, creation date and links used to activate or deactivate all clients in a lot. Before activation or deactivation the system asks about confirmation. It is possible to remove the selected lot by clicking the Delete button above the list. If more than one is checked the checkbox system will remove all checked. Removing lot will remove all clients belonging to it and operation cannot be undone. Creating lot

Creating lot of clients is divided on few section. Every such section has fields used to define parameters for generated clients.

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Lot's propperties:

1.

a. Descripition - lot name which allows to identify group of clients. In clients list it is possible to filter clients using this name. From list of available lots it can be activated, deactivated or removed by selecting this name and choosing appropriate action. Lot name should be detailed to be to easily recognized in the list of other lots Number of new clients - number of clients to create b. Starting serial - this number will be used to identify every client created in lot. It can be used as card number if logins or passwords are printed on card. In export of lot this number will be available and used later for priting or client identification. If there is new lot and cards number don't start from 1 than serail number can be set with any value and new serials will start from given value. c. Users - type of clients to generate. Changing type of client changing also other section enabling or disabling option available for different kind of clients.

Logins and passwords

Options defined in these section are the same but they are used for login and password generation for new clients. 1.

a. Number of characters - number of characters used to create login or password. Type of characters used to generate is defined below. b. Starting characters - every login or password can start from some initial starting characters. It is used to easily identify all clients or can be used to set dialing plan with only one entry to all these clients. Such scenario is described here. Starting characters cannnot be too long in comparison to Number of characters. For example setting login length as 5 and starting characters as value 7777 will allow only to generate 9

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different logins. Depending on client type the login or password must be unique so if there are any other clients created already it will narrow possible values. Logins and passwords are generated randomly. Every character used in login or password is randomly generated and its type depends on which option is checked. c. Use numbers d. Use up cases e. Use low cases If more than one option is checked the system will generate it as a mix of different characters. For GK clients it is good to create logins as numbers only so later it is easy to set dialing plan for them without any number modificiations. For IVR clients password is used as pin to log to system so it must be defined also as number because letters are not possbile to enter from phone keypad. Sequential generation - allows to generate login or password sequentially. Below there is Starting number and step. During client generation it will start from starting number and every new login or password will be increased by step value. If there is starting characters set it will add generated value to it. It wont be added as number but as concatention of characters for example starting characters set as 1000 and starting number as 3000 will create first client as 10003000 and not 4000. Starting number Step

Client's properties: Values defined there are the same as used when client is added or edited manually. The only difference is that value set there will be used to create all clients in this lot. Some fields in this section are activated or deactivated depending on the kind of client chosen to generate. Tariff Chose tariff according to rules Funds Dest. dialing plan prefix Dest. tariff prefix Dest. caller ID prefix Src. dialing plan prefix Src. tariff prefix Src. caller ID prefix Connection settings: Allows to define special properties used by clients of chosen type. Supported codecs Create lot can be assigned to reseller by clicking right mouse button on selected lot. Context men will appear with command "Add to reseller". After choosing reseller lot and all clients belonging to it will be assigned to given reseller. Assigning lot in this way is not typical way and it will not cause changing "Clients limit" value for reseller. Normally resellers should create their lots from reseller system VSR. Lot export is available by clicking Export button above list of lots.

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CSV comma delimited file is used as output format. Such file can be opened and modifed by Excel or notepad. During export operation there is progress window available presenting current status of operation and when option open file after finishing is checked the system will open exported file automatically when finished.

Currencies management Description This feature allows to assign different currencies to clients. For example, if VoipSwitch owner is charged in USD and his clients want to be charged in EUR, one may keep USD as base currency but assign clients EUR. One thing is very important to work properly with currencies in VoipSwitch. Tariff assigned to a client and payments added should be considered currency defined for him. Rates in tariff are added only with value and only assinging them to clients will define what currency and ratio is used to calculate cost of a call. The same goes for payments. Amounts added must be connected with currency defined for every client. Currencies are not supported for any level of resellers or costs calculation for termination devices. Only based tariff can be used to calculate their cost. All tariffs assigned to resellers or termination gateways must be in the same currency which is treated as base. Future browsing calls made by clients in VSM, VSC or VSR will show value made in base currency. Clients logging on the web and portal will be able to see these values modified by ratio defined for currency. Values taken from calls are multiplied by ratio assigned to currency defined for clients and taken for browsed date.

Currency definition First thing to do when you start working with tariffs is adding currencies. Every currency has defined number of ratios assigned with dates. It is important to keeping them up to date.

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Adding ratio values In the main currencies window you can add or edit a list of managed currencies. When you click currency name, the dialog box with ratios for given dates appears.

Ratio has the function of dividing the cost of a call before it is saved in the database. This means that all costs for resellers, clients and termination devices have the same currency, which will allow to calculate profits. Before the cost of a call is displayed for a client on the web it is multiplied by ratio saved with the call, and the client can see proper value on the screen. Changing ratio for previous dates will not change cost of calls made by clients. Changing ratio for the last day will cause new calculation for calls made after this change. Ratio used to calculate the cost of a call is stored with every call and it cannot be changed after finishing the call.

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Advanced - currency proccessing 1. Values used in tariff or in payments should be in a currency assigned to a client. Tariff definition has no difference for different clients. Only currency assigned to client will cause using rates differently. The same is for payments. 2. When client is connecting to VoipSwitch, the remaining time is checked. Tariff and amount of money on client account is used to calculate the cost of time remaining. If tariff has appropriate rates and amounts are defined in the same currency, all is valid and client will be disconnected properly upon reaching 0 amount. 3. After finishing a call the cost of reseller or cost of termination is calculated without any change. There is a difference for a client who has different currency. Before the cost of a call is saved in the calls table it is modified by appropriate ratio taken from the currencies table. Doing this recalcualtion will save cost for client in the same currency as for other costs. It will allow to estimate profits properly.

4.

A client after logging to his web pages will see costs taken from calls table but multiplied by ratio. After saving cost of call for client ratio is save in every calls record and later used to show values in client currency. Browsing calls in VSM,VSR and VSC will show results without any multiplying.

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Client's export,import

VSM and VSC export and imports Client's information using comma delimited CSV format without column names. Eeach row represents one client definition. Note: In VSM - user could select which column may be exported - but this probably cause problems in importing this data back to VSM nad VSC, because only all-columns exports could be imported back. Client format: test,123,3277362,173,235.0000,DP:;TP:;CP:,-1,-1,0 Field meaning (counting from 1): 1. Client login 2. Password 3. Client type - value set there is coding option available for client definition like codecs, connect immediately and others. 4. Tariff ID in system (or Tariff Interstate ID) 5. Account state - is internal number assigned to every tariff created in system. It is not presented anywhere in the system and can be seen only in export file. 6. Tech prefix - values coded here are used as tariff prefix, dialing plan prefix and caller id prefix. This value is coded from appropriate text boxes in client definition 7. Reseller ID in system - internal number assigned to reseller of first level. This number is not visible in system. 8. Intrastate Tariff ID from system 9. Calls limit - it stands for calls limit value limiting number of concurrent calls being accepted from defined client.

Fig. 6.1.1: Export client's operation with visible dialogs: a) Performing task progress (default dialog in VSM for long tasks), b) Select columns which You wan't to save to file.

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As described above some fields are difficult to create by someone who wants to import clients. It is recomended to export first one or few clients with proper definition. Later using Excel it can be modified and multiplied. The value of some fields and others can be filled with logins and password or account_state values. The file can be saved from Excel using CSV file format and imported using VSM or VSC application. In the future it will be availble to import clients using special form will to fill coded values.

CallShop clients export

Format has some differences fom standard client's modules export: Client format: ntc,123,61,0.0000,-1, Field meaning (counting from 1): 1. 2. 3. 4. 5. 6.

Client login Password Tariff ID From system Account state Reseller ID from system Tech prefix

In Callshop there is no possible to assign Interstate/Intrastate tariff, so this field is not supported by export too. Because of standard-callshop file format differences there could be problems with interchange data between callshop-other client types.

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Prefixes This page last changed on Feb 04, 2008 by rashid.

Prefixes are strings normally built from digits but they can also have some characters. They can be modified in different ways. Below is a list of available ways to modify them: • • • • •

adding characters at the begining removing matching characters from the begining adding characters at the end validating length of prefix being sent replace prefix sent from a client with explict value

Examples of use • • • • •

Removing leading zeros from dialed number Adding zeros to dialled number Change gateway for the same country depending from client calling Redirect all calls from client to Voipbox scenario Advanced number manipulation

Such conversions can be used separately or together. Prefixes can be modifed directly or through helping dialog. You can find the helping dialog after clicking the button near the rule definition textbox.

After clicking this button the helping pop-up window will help you create the rules.

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Helping window allows to set informations: •

° Forward from client is equivalent to an empty prefix field and it will forward to prefix exactly how it was received from client.

•

° Always send will fill the prefix field in the format !123 where 123 is the number desired to be sent. That means the entire prefix received from client will be always substituted with value you define here.

•

° Change will fill the prefix field in the format "X->Y|Z" where X is the prefix field from the helping window, Y is the change to field and Z is the suffix. It means that if the prefix received from client starts with X then replace it with Y and add Z at the end of the entire prefix. - prefix - value defined in this field will be replaced with value set as to. If prefix does not match the begining of number no action will be taken. To remove first characters from number sent to should be empty and then prefix will be replaced with empty - in face removed. - to - value which will be added the begining of number. It will be added only when prefix field is empty. If not value from prefix will be replaced with this value. - add sufix - it will add this value at the end of the sent number.

•

° Required number length - lets the user define a required string length that will be accepted

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by VoipSwitch. In a scenario when the string does not meet the defined requirements such call will be rejected. It can be used to allow only calls of valid length or from valid callers id to be processed. Examples of modyfing prefixes

Removing leading zeros from dialed number

example 1 For better understanding of this mechanism consider that the client (in VoipSwitch, Clients are originators) sends a call to VoipSwitch. First, VoipSwitch will want to know how to bill this call depending on the destination so it will have to search in the tariff allocated for this client's matching rate. Here comes the help of Tariff prefix. For usual cases when the client is dialing exactly with the prefixes you have in tariff you will leave the Tariff prefix empty. But when the client dials with 00 and in your tariff you have only prefixes without 00 then you enter a replacement rule in Tariff prefix field like 00->. This will cut the 00 if exists before the number is sent to Tariff to match a rate. The same rule can be set for Dialing plan prefix when the client dials with 00 and in Dialing plan we have only routes for country codes prefixes. In this case we can fill the Dialingplan prefix field with the value 00->. That means we will cut 00 (replace 00 with nothing) from numbers dialed by client before we sent them to the Dialplan routing. And this is even better because the client can dial either with 00 or without 00 while this replacement rule will cut only if 00 exist at the beginning of number. Of course you can leave Dialingplan prefix empty if you have routes in dialplan exactly for what the client is dialing. Separate rules can be created for each client by providing a different Dialplan prefix.

Adding zeros to dialled number

example 2 Also you can consider the case that your client always dials without 00 while in your tariffs you have all the prefixes starting with 00. In this case you fill the Tariff prefix with the value 00. So 00 will be added in front of all dialed numbers received from client before they are sent to Tariffs to match a rate. You can imagine how useful is this because you will not be forced to create one tariff with 00 and another without 00 with same rates, and then from time to time to be forced to update both. Now that VoipSwitch found the rate and knows how to charge this call it will try to send it to Dialingplan to find a matching route for the dialed number prefix. And here again we can have a lot of help from the Dialingplan prefix field. Before the number is sent to the Dialingplan for routing we can add the same rule or different depending on how dialing plan is built.

Change gateway for the same country depending from client calling

example 3 For example we want to route all calls from gateway B through destination Termination 1 and all calls

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from gateway A, to the same country, through Termination 2. So we will give to gateway A a Dialingplan prefix like $ sign and to gateway B a Dialingplan prefix will be empty. Then in main Dialing plan all we have to do is to create routing rules for telephone numbers starting with $ and others just starting with country code. We will know that calls with $ are coming from gateway B and those with country codes are coming from gateway A. So we can route same country to different terminations gateways using the Dialplan prefix as an internal tech prefix. It is important that before sending number to destination gateway this special prefix $ should be removed and it can be done by rules for modifing client's data. It should be noted that adding these special prefixes will not cause changing number when rate is being looked for in tariff.

Redirect all calls from client to Voipbox scenario

example 4 When working with calling cards services it is neccessary to authorize and redirect calls coming from DID carrier to PIN scenario available in VoipBox. Clients are informed about some number which is redirected from this DID carrier to VoipSwitch and when they call it they can hear asking about a pin. To be sure that any other number will not be sent from this carrier to VoipSwitch and by mistake connected through VoipSwitch we can define special dialing plan prefix. On every number sent from this gateway we can add special prefix or character and then in the main dialing plan add entry with the same value as added prefix. It will assure that no matter what number is sent from this client it will always be connected to defined scenario or more general destination. As a dialing plan prefix for this client it can be set any value but it shouldn't cover existing country or region code used by other clients. Some examples are #,$ or 7777 etc.

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In the main dialing plan the same prefix should be defined as presented on the figure below.

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To be sure that even if other such prefix is defined in dialing plan and a call will not go there it can be selected option Don't jump

Advanced number manipulation

It is possible to manipulate dialed number in an advanced way. You may add, remove or change a place of every digit or set of digits. Rules: • incoming number can be split into parts (named s1, s2, s3 ...) Illustration

As you can see number 48600789456 is split into 3 exemplary parts. • every part of incoming number has fixed length (defined as : s1{4}, s2{3}, s3{1}, ...) • split parts of incoming number has to cover all digits of incoming number (either more or less) Illustration

As you can see exemplary split is not covering all digits, so can't be used. • transformed number doesn't have to contain all split parts of incoming number. • transformed number doesn't have to use whole split part, it is possible to take only few initial digits from a part. • split parts can be used in any order with prefix, suffix after them. Do not use additional "|" as suffix separator. When using advanced number manipulation with split parts of number, you cannot use other regular number manipulation (changing dialed number, add/remove prefix, add suffix). However,

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such manipulation can be done with this mode. Some examples are given below. example 5 (adding prefix) Adding prefix (00) s1*->00s1* exemplary number dialed: 48600789456

will be send as:

0048600789456

will be send as:

4860078945600

example 6 (adding suffix) Adding suffix (00) s1*->s1{*}00 exemplary number dialed: 48600789456

example 7 (changing dialed number) Adding suffix (00) s1*->48501456456 exemplary number dialed: 48600789456

will be replaced:

48501456456

example 8 Adding extra digits inside dialed number s1{2}s2{3}s3{6}->s1{2}4s2{3}5s3{6} exemplary number dialed: 48600789456 will be send as:

4846005789456

example 9 Changing order of dialed number. First two digits are moved 3 places to the right. s1{2}s2{3}s3{6}->s2{3}s1{2}s3{6} exemplary number dialed: 48600789456

will be send as:

60048789456

Advanced number manipulation works also for PIN Prefix (for GW Clients). Prefixes are used also when modifying number being sent to destination gateway. There are different entries, but rules for conversion of number or caller id send there are the same. Below is an example of few rules with explanation. DN:#->;

removes any appearance of # in dialed number (DN) in front of number ( -> stands for replace, in given example # -> replace with empty )

DN:234->997;

replaces dialed number (DN) starting numbers 234 with 997 before sending to destination

DN:77;CP:!3000;

adds 77 in front of dialed number and modifies caller id to always send 3000 value ( ! stands for 'always send' )

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3.0 Destinations This page last changed on Feb 24, 2008 by admin.

2.0 Destinations Every call coming to VoipSwitch is first authorized by proper definition of clients. Then the dialed number is checked and depending on the dialing plan rules it is set to specified destination. There are 4 types of destinations where VoipSwitch can send calls. Gateways Gatekeepers GK, PC2Phone, Common clients VoipBox (IP IVR)

2.1 Gateways In this section you have to define the termination gateways where you will send calls. VoipSwitch will send the calls to these gateways in direct mode (IP to IP).

Fig 2.1a Gateway definition VSM

Fig 2.1b Gateway definition VSC

Every option available to set for gateway is described below: • Description is a label for the terminating gateway. • IP number is the IP address of remote terminating GW. Instead of IP it can be set when domain name is like sip5060.arbinet.com

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• Port on remote gateway where to send the calls. Standard port for h323 protocol is 1720 and for SIP 5060. You have to change the port manually when you change the protocol. • Active sets the gateway active or inactive. • Calls limit sets a limit of maximum simultaneous calls that Voipswitch is allowed to send to this terminating gateway. Zero means unlimited calls. • Supported codecs defines codecs accepted by the remote gateway. • H323 device or SIP device to select the protocol that Voipswitch will use when sending calls to this gateway. • H323ID and FastStart are options that can be set when you select H323 protocol. H323ID can be required by your termination carrier to be sent for authentication. If not required it is safe to be left blank. FastStart is a specific h323 protocol feature that enables faster call connection and advanced in-call options like call on hold and forwarding. You have to ask your carrier if his terminating gateway accepts this feature. When you select SIP protocol you will be presented with Username and Password fields. Set them according to the terminating carrier requests or leave them bank. • Early H245 • Calculate cost and Tariff when this check box is selected there must be tariff chosen from combo box. This tariff will be used to calculate the cost of connection after finishing every call terminated using this gateway. Tariff used there is defined in the same way as any other tariff in VoipSwitch. It should be named differently than tariffs used for clients. Cost of calculation allows later to compare bills received from carrier or to see profit for calls made by clients.

2.2 Gatekeepers VoipSwitch can log to gatekeeper or registrar by itself and then send calls there. All information used to register should be provided there. Useful function is LRQ which allows to negotiated with Gatekeeper new ip address and new format number. This option must be supported by gatekeeper and VoipSwitch will handle it. After you create the GK/Registrar account you can go to the main VoipSwitch window and click the button Relog to gatekeepers from Gatekeepers sub-window to make VoipSwitch try to register immediately. Properly defined and configured gatekeeper is marked on VoipSwitch in blue color. If there is any problem it is marked in red.

Fig 2.2a Gatekeeper/Registrar definition VSM

Fig 2.2b Gatekeeper/Registrar definition VSC

• Description field is a label for the termination account. • IP number sets the remote GK or Registrar IP address. • Port where to send the registration request (usually 1719 for h323 Gatekeepers and 5060 for SIP Registrars). • Time To Live in seconds. It sets the amount of time until Voipswitch will check again if the remote GK or Registrar still accepts calls. Better to set this value smaller or equal than the value set on remote side. • Supported codecs accepted by remote side.

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• Gatekeeper (h323): H323 ID, e164, GK name, FastStart - consult your carrier about these settings. If not required leave them blank. But you should set at least GK Name and FastStart. • Registrar (SIP): User name, Password, Domain user, Domain - consult your carrier about the values in these fields.

2.3 Clients defined in VoipSwitch Any number set in VoipSwitch can be redirected to clients logged in VoipSwitch. Types of clients which can be used are : • Pc2phone • GK/Registrar • Common clients After selecting the type of client specified login should be chosen from the list of clients.

After assignmenta call coming to this number will be sent to such device or dialer. No matter what IP is used to login or whether a device is logged from behind NAT ( SIP protocol only ) the call will be sent properly. Sometimes it is required to modify number sent to client device to change according to rules. On web pages available as Web or Portal module client can define what should be done when his device is not logged, busy or not answering. Different types of redirection are available to Voicemail or number. It can be defined to charge client for calls redirection depending on his tariff. It is possible to charge clients for answering calls sent to them and client tariff will be used to calculate cost. For bigger number of clients it is possible to define redirection for all clients using just one entry in dialing plan. Every such client should have the same beginning.

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2.4 IP IVR ( VoipBox ) Last type of destination is VoipSwitch IVR system. Client connecting to such destination can hear voice depending on the assigned scenario. Available scenarios are described in Voipbox manual. To list only few example it can be used for playing account state information, getting clients PIN number in calling card services, asking for number, etc. All details are described in IP IVR module section.

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4.0 Dialing plan This page last changed on Feb 24, 2008 by admin.

• • • • • • • • •

4.1 Base informations 4.2 Calling modes 4.3 Load balancing 4.4 Rules for modifining clients data 4.5 Automatic calls redirection to group of clients 4.6 Special properties 4.7 Time span 4.8 Importing and exporting data ° 4.8.1 Export dialing plan ° 4.8.2 Importing dialing plan

4.1 Base informations Dialing plan is used to route calls to destinations. Rules are based on dialed numbers. First characters of numbers are named prefixes. Every prefix is assigned with a destination. VoipSwitch searches for matching prefixes and tries to send call to the most detailed ( longest ) prefix. For example, when 48 600 316 151 number is dialed and prefixes 48600 and 48 are defined in dialing plan system will try first 48600. If gateway defined for first matchin prefix is not connecting, gateway defined for less detailed will be used. The same prefixes can have different priorities to set order of choosing them.

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This part of manual describes rules for creating dialing plan entries and available options.

4.2 Calling modes It is used to define special properties when passing a call between origination ( client ) and termination side. Modes chosen depend on protocol used by a client and destination. Modes available for H323 client calling to H323 destination: • Proxy all, connect independently - origination and termination endpoints do not see each other, the VPS connects independently with each endpoint then conferences them together. • Proxy all, forward call signaling and H245 signaling channels - the signaling and media will still be passed through VoipSwitch as in first rule. The difference between this option and the previous is that the call setup received from the client is sent to the target gateway. So the two endpoints can use more codecs if they both support them (even if VoipSwitch doesn't support it). Also, information coming from a client through h245 channel is forwarded directly to the termination gateway. So H245 tunneling can be used (if both endpoints support it). • Proxy call signaling and H245 channels, no media proxy - only signaling information and H245 channel are passed through the switch, media packages are sent directly between endpoints. • Proxy call signaling only, no H245 and media proxy - in this mode only signaling information is passed through the switch. All the rest are flowing directly between the two endpoints. Modes available for SIP client calling to SIP destination: Modes available for h323 client calling to SIP destination and from SIP to h323 ( changing protocol ):

4.3 Load balancing This function allows to set percentage of calls being sent to different destinations for the same prefix. It is useful to split traffic between gateways for the same country. Any number of entries in a dialing plan can be set in this way but they must fulfill special requirements: 1.

a. Telephone number is exactly the same for each entry, b. Priority is exactly the same for each entry. c. Summary of balance share value for all entries must equal 100.

In case when one wants three gateways equally balanced one must enter 33/33/34 Balance share for those dialing plans (as shown below). The balance share does not have to be equal for each entry, but their sum has to be 100.

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For better finding entries with defined load balancing all of them are selected in different green color.

4.4 Rules for modifining clients data

This field is complex and allows modifying different call settings last time before the call is sent to the termination gateway. At the end of this field there is a button with 2 dots. This will open a helping window that will guide you through the possible settings for this rule. Information available to be changed: •

° Dialed number - allows changing number. ° IE Display and IE Calling party number are 2 fields from H323 protocol that refer to caller ID information. If you want to modify the caller ID sent to termination you should modify either one or both fields depending on termination provider (some accept first field others work with the second field). ° H323 ID - sent to the termination GW can be modified or defined here as well. All these h323 field changes will be taken in consideration only when routing h323 calls! If you want to modify the caller ID for a SIP call first route it as h323 to own and then forward the information to termination GW as SIP!

Rules definition on how every string is changing are described with details in section prefixes.

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Very common usage of field Dialed number is to add some prefix before sending a call to the specified gateway. Some carriers require it for authorization or different billing. VoipSwitch owner can have clear dialing plan with real country codes. Using these rules it can be modified. In given example number 31798804370 is modified by adding in front prefix 77678 so on destination gateway 1233 will be received number 7767831798804370 Other common option is to replace number dialed by client to number expected on IP phone. If destination is IP phone logged to VoipSwitch as GK Registrar client than in most cases it will respond only to the number being the same as login. If we want to redirect some DID number then rule must be defined as show on screen. In given example number 33172898106 is replaced with login name which is sipura1 before it is sent to destination IP phone logged with login sipura1. Every device has special field used as number to which it responds. Below is a list of different devices with these fields marked. Sipura, Cisco ATA, Option disable "folow me" for DNIS mapping is used to disable follow me for entries with any map to dnis option. Checking it will block redirection for pc2phone, gk or common client being called.

4.5 Automatic calls redirection to group of clients • Map DNIS to GK/Registrar accounts • Map DNIS to PC2Phone accounts • Map DNIS to common clients accounts All these Map DNIS to... features will automatically route the calls to the GK/Registrar, PC2Phone or Common client account that will have the Login as the dialed number. For example, to route calls internally between all your pc2phone clients all you need to do is to create the pc2phone accounts with distinct numbers as Login name and then add a Dialingplan rule having Map DNIS to PC2Phone accounts enabled. When creating this Dialingplan rule it must be set this distinct number Telephone number and any Destination because it will not count in the routing process, however it must be set with some value.

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When this option is used Follow me feature is automatically turned on.

4.6 Special properties • Prefix not allowed - used to block any call coming to a number starting with such prefix. Even if there are other prefixes matching the dialed number rerouting will not be made. Common practice is to block special expensive numbers. • Route disabled - For some reason ( expected gateway inaccessibility ) we can disable prefix without removing it from the dialing plan. Such prefix will not be used to send calls and later it can be easily restored. • Follow me This option is used only when assigned with terminations as PC2Phone, GK or common clients. Client can define on web page that depending on specified reasons call should be redirected. It means that a call can follow if a client is not answering. Instead of phone number a client can define that it should go to voicemail. • Do not jump - when you have multiple rules for same prefix you can enable this option to stop the hunting when the call will fail through the current rule. • Do not announce time - option used to stop announcing time when a call is made to this number. Announcing the remaining time is one of the features of IP IVR module. For some service numbers ( account state information, recharge scenario ) time announcement is not required and can be blocked using this option. • MediaWaitForConnect if enabled will make VoipSwitch to instruct the origination device to generate fake ring tone while waiting for connection. Not all devices can generate fake ring. But Cisco ATA and others can, so this helps sometimes when remote gateway doesn't send proper alerting. • Allow Voipbox to send media before client - used to send recording from voipbox without sending connect message to client. It is used commonly with callback service. DID number used to activate callback can be set in dialing to Voipbox and Play file scenario. This scenario is playing recording as no answer voice. Voice is played to client but connect is not returned so there is no charging for such call.

4.7 Time span Every prefix defined in the dialing plan can have set time or day of week when it will be used. Different destinations can be used in different time.

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Exporting of dialing plan positions is available in VSM and VSC web config. After choosing dialing plan position from a menu tree the import export buttons will appear in the upper right corner of the screen.

fig. Export buttons for exporting dialing plan positions Clicking on Export button causes exporting all dialing plan positions. If filter is applied only filtered records will be exported. Format used for export is coma delimited CSV files. Such file can be opened in Notepad or even by Excel for further modifications. System will ask then about location and name of export file and then it will be completed. Exported columns order Telephone number

Priority

Route_type

Id_route

Tech_prefix

Call_type

type

'#'

0

0

65

'DN:#->;'

16

0

Telephone number and priority are self-explanatory and the same titles can be found in a form when editing dialing plan position in VSM or VSC. Route_type and id_route defines where calls for given number will be send. Route_type description: 0

External gateways

Gateways

1

Internal gatekeepers

ClientsE164

2

External gatekeepers

Gatekeepers

3

PC2Phone clients

clientshearlink

4

VoipBox ( IVR )

scenarios loaded dynamically in voipbox

5

Common clients

clientshared

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Tech prefix stores value for part defined in VSM as Rules for modifying client's data. This text value has coded conversion rules for the dialed number, caller ID, h323 id before sending them to destination from VoipSwitch. It is quite complicated to manipulate directly those values but anyone interested can define some test entry with valid conversion and later use it in these files ( for import purpose ). Examples of string manipulations are defined here Call type value is binary coded and it defines dialing plan mode. Depending on which protocol is using specified route ( SIP or H323 ) this value is differently decoded. It is not recommended to modify it manually. Type has coded definition of values defined in VPSconfig as Special properties, Don't jump, MediaWaitForConnect. It shouldn't be modified directly but rather copied from existing row.

4.8.2 Importing dialing plan

Of course file with export from the dialing plan can be used as import. But there is a feature which allows importing incomplete rows from file. The only one required field is telephone number. If other fields are empty ( for example '4877',,,,,,,) the system will present a form to fill missing values before using file for import. This form is the same as the one used with adding or editing dialing plan positions. Some parts are hidden or displayed depending which fields are in import file.

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fig. Filling missing information in dialing plan import Only first row is checked and other rows are imported with values filled using this form. Example of such incomplete row '4877',0,,,'DN:9889',,

This row will cause the system to ask about destination device and call type. These values must be picked up in a form. Other rows in this file will have the same values except columns filled with not empty values like telephone_number, priority, tech_prefix where every value will be taken from appropriate row. Only setting two comas without any character between them will cause asking about missing value. Space character or two apostrophies '' between comas won't be taken as empty.

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5.0 Tariffs This page last changed on Feb 21, 2008 by admin.

• Base informations • Parameters for tariffs • ° Resolution ° Minimal time ° Surcharge time ° Surcharge amount ° Time span ° Tariff multiplier ° Tariff addition • Tariff prefixes • ° Grace period ° Resolution ° Minimal time ° Rate multiplier ° Rate addition ° Disable prefix ° From day, to day, from hour, to hour • Importing tariff prefixes • Changing tariff for clients • ° Caller ID ° DNIS number ° Using NPA function ° Tariff comparer • Calculating cost of call by VoipSwitch

Base informations Tariff in VoipSwitch defines a set of paremeters used to calculate cost of a call. Every tariff is built of prefixes with assigned minute price for them. Few parameters can be defined to whole tariff and some of them can be defined for specific prefixes. All tariffs are defined in one place and later a tariff can be used for different purpose. Tariff can be assigned to: • • • •

any client defined in VoipSwitch to charge him for calling gateway, SIP proxy or gatekeeper to calculate cost of termination reseller of any level to calculate cost for him for special usage like Tariff to DNIS or Tariff to ANI

Every tariff is defined the same way and only assigning them causes different usage. A call will be connected only if the prefix of the dialed number exists in the tariff. All the dialed numbers without matching prefixes in tariff table will be rejected. Prefix must exists in tariff assigned to a client, reseller (if client is assinged to reseller) or in tariff assigned to gateway if calculating cost is set. List of defined tariffs can be accessed by clicking Tariffs node in VSM or in VSC.

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After you click it, a list of tariffs is presented on screen.

Parameters for tariffs Every tariff is identfied by name and set of parameters used to calculate cost of calls.

Resolution Every rate assigned to prefix is for one minute. Tariff definition allows to charge clients for shorter periods from 1 second to any number of minutes. Resolution is a parameter which is used for that. Value of resolution is in seconds and defines when part of a minute price should be added to cost of a call. For example when it is set with 6 it means that every 6 seconds one of tenth minute price will be added to cost. As resolution it can be set at any value but for clear calculation it should have a value which divides 60 without rest.

Minimal time

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This value is used to define minimal time for which a client will be charged after connection. Even if a call is connected and disconnected after few seconds a client will be charged for this time. Resolution steps required to cover minimal time will be used to calculate cost. If resolution is 6 and minimal time is 40 then minimal time will be 7 billing steps because 7*6=42 seconds.

Surcharge time This value defines time in seconds which can be charged by surcharge amount. Remaining time of a call will be charged using other rules described in minimal time and resolution. When surcharge amount is set at 0 then time defined in surcharge time will be free for client. For example when surcharge time is defined as 10 and duration of call was 60 then surcharge amount will be added to cost of a call and the rest of 50 seconds will be calculated using minimal time and resolution.

Surcharge amount Amount added to every call for begining of each call is defined by surcharge time. If surcharge time is set at 0 then it will work as connection fee for any connected call. After the Surcharge time expires the billing will start as if it is the beginning of the call. For example if the Surcharge time field is 300 seconds and Surcharge amount is 0.1 then the first 10 seconds of each call will be charged with 0.1 and only then the normal billing will start.

Time span When this option is checked it allows to define different rates used for different days or hours. Unchecking this option makes setting tariff easier and will speed up cost calculation. If you would like to setup some rate for hours crossing the midnight (ie 8.00PM - 6:00AM) you shouldn't add this rule in one stage. Insted of this add two rules (first one: 8:00PM - 12:00PM, and second one: 0:00AM - 6:00AM).

Tariff multiplier Client can see his tariff rates and prefixes on the web page after logging. Using this option allows rates visible by a client to be changed during call. Until a client recalculates cost of a call manually he will not be aware of it. Rate multiplier is changing cost of every rate by multiplying it by this value. Examples. If it is set 1.1 every rate will be 10% higher than client can see on the web. Value 0.8 will decrease every rate by 20%.

Tariff addition Similar to tariff multiplier but instead of multiplying it adds some value to every rate.

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Prefixes defined for a tariff are presented on screen after clicking tariff name.

Prefixes are assigned with rates and descriptions. Tariff can have any number of prefixes defined in it. The same as in dialing plan longer ( more detailed ) prefixes matching dialed number are taken first before shorter. Examples

Client calls number 48 600 316 151. If in tariff are defined prefixes 4, 48, 486 then prefix 486 will be used to calculate cost of this call. If rate for 486 and 48 are the same then 486 can be removed. Having less number of prefixes in tariffs can speed up processing. Common practice for setting a tariff for given country is to define general number for a country and then only define more detailed prefixes with different rates. Descriptions defined in every prefix should be filled properly because they are used later in detailed billing, on client web CDR page or in Reports. The same description can be used for different prefixes and later can be used for more general grouping. For example 486 and 485 are Poland mobiles however with different cost. Later in summary we can see it grouped by description and see how many calls went to polish mobiles. If price is the same for many similar prefixes it must be considered if it cannot be replaced with one more general prefix. Tariffs with smaller number of prefixes are easier to manage and with higher traffic can be processed faster. Examples

Having prefix 480,481,482,483,484,485,486,487,488,489 set with the same rate 0.1 all of them can be replaced with one entry 48 with rate 0.1. If one of them has different rate then it can be set 48 with 0.1 and 486 with 0.4 and then only number matching 486 will be billed differently. For every prefix some parameters can be set to modify calculation of cost. Some parameters are the same as for tariff. If such parameter has value higher then 0 it will be used instead of the one defined in tariff. It allows to set different value for specific prefixes.

Grace period If any connected call is shorter then this time it will appear in CDR with 0 cost . Calls longer then this time will be billed normally with the whole duration including this grace period.

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Resolution See resolution defined for tariff.

Minimal time See minimal time defined for tariff.

Rate multiplier See rate multiplier defined for tariff.

Rate addition See rate addition defined for tariff

Disable prefix Any number which matches this prefix will be blocked from processing. System will reject such call. Similar to option used in dialing plan used to block some prefixes.

From day, to day, from hour, to hour Values set there defines when a prefix will be used. It is possible to define from which day of a week to which day and between which hours this prefix will be valid. Below is an example of how it should be set to have offpeak and onpeak rates. Example

Importing tariff prefixes It is possible to import the tariff rates from a CSV file. For this you will have to prepare the file in the following specific format (order of columns): Prefix, Description, Rate, From Day, To Day, From Hour, To Hour, Grace Period, Minimum Time, Resolution Be sure you don't have column names in the text file, and the fields are comma (,) seperated. The file should not contain comments or column headers and data should start from the first row. Usually you can work this rate file in Microsoft Excel or Open Office and save it as CSV. The file should look like this:

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93,AFGHANISTAN,0.2243,0,6,0,2400,0,0,0 4795,NORWAY - zz Mobile Teleno,0.1938,0,6,0,2400,0,0,0 47960,NORWAY - zz Mobile Teleno,0.1938,0,6,0,2400,0,0,0 You can also import tariffs from files that had tariffs exported before. Please note that if you are importing Tariff using Web Config the fields in the CSV must be seperated with ; not (,). When you have the file ready, upload it on the VoipSwitch server, and from your tariff settings in VoipSwitch Manager or VoipSwitch Web Config press Import button.

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You will be asked to select the text file. Some fields can be empty but if you won't, you can make them useful by putting the data in. VoipSwitch Manager will stop the importing process and announce the error if the process fails. The records from the file will be added to the existing records in the tariff. You must remove the existing records before importing if you want to replace them. There is a Remove all button that will delete all rates in that tariff for your convenience.

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Changing tariff for clients Client can have one tariff assigned to him but there are a few ways to change such tariff depending on information sent to VoipSwitch. This change can be done because of

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This mode can be used only with IVR clients or common clients working as IVR. This function is named as 'Tariff to ANI" and is used only while providing the 'calling cards' service. Every reseller can set different rules for his clients on how tariff should be changed depending on the number called. Tariff to ANI function is changing tariff assigned to a client to some other depending on caller id coming to VoipSwitch. It can be used to differentiate cost of calls when client is calling from abroad. Tariff to ANI detailes description

DNIS number Using this feature a tariff is changed depending on the number called. It is used in calling cards service. Typical scenario is to set different number for calling cards and then depending on different tariffs it will be used to charge clients. One number used to call to IVR system can be toll free and other charged for every connection. Only by using this feature we can differentiate tariff used to charge the same client. Detailed description how to set it is described here.

Using NPA function This function allows a user to change the tariff depending on the number that is dialled and the number that makes the call (using Caller ID). This function is exceptionaly usefull in the USA, where there is a need to change the tariff not only on the basis of the state called but also on the basis of the state the call is made from. A simple comparison of numbers is not enough, so there has to be a special table, that links places with codes. In the Others section one can find a Numbers Table with a location (name) linked to it.

Procedure of tariff change.

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1. The user calls a number that begins with a certain string of numbers. 2. The system checks which localisation has such a number assigned. 3. After finding the location the system compares the beginning of the clients Caller ID and if it is assigned to the same location it uses the Intrastate Tariff. 4. If there is no match the tariff used is Interstate. Definition of Interstate and Intrastate tariff is availble on dialog which appears after clicking the „Choose tariff according to" and the Rules button. After clicking it one can configure the Tariffs for the client in a new window.

Example

Tariff comparer Option is available under tariff node.

Option used to compare 2 tariffs. Working with traffic and clients requires many tariffs, prefixes and rates. Mistake in one rate can cause big loss for VoipSwitch owner so it is very important to check every tariff before assigning to clients. Tariff comparer is one of features available in VSM application. It is possible to chose two tariffs and define criteria for comparing tariffs. If one tariff should be lower, higher, equal then any rate in the other. It is possible to define how much bigger or smaller it should be. Matching prefixes are listed in output window and can be modified. Tariff comparison is more complex

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and works in a similar way like finding prefix in tariff made by VoipSwitch. If the prefix in tariff is not the same the best matching entry is taken to comparison.

Example shown in figure above displays only prefixes from tariff TestTariff which are higher than 20% of appropriate prefixes from TestTariffHigh. Value from Tariff 2 voice rate is used to multiply prefixes from Tariff 2. Examples To find prefixes from Tariff 1 which are bigger than in Tariff 2, a voice rate should be set with value 1 and operator as >. Tariff comparer should be used every time when prefixes are changed. Comparing tariff used for clients and used to calculate cost can find mistakes and avoid money loss if it will be noticed later after some calls. Tariff assigned to resellers should be also compared with cost tariff to earn on every prefix. Make sure all prefixes assigned to reseller tariff are higher than in cost tariff.

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Tariff to ANI This page last changed on Feb 24, 2008 by admin.

Typical applications

The function 'Tariff to ANI' enables to change the tariff for a client according to the telephone number from which the connection is being made. This function is used mainly while providing the 'calling cards' service. Clients call from the telephone network (PSTN) to the VoipSwitch system. Then the connection is transferred to the IVR system, where the client becomes authorized by aPIN number. Sometimes the VoipSwitch owner pays for the connection from the PSTN network to the VoipSwitch system. The cost of such a connection depends on the type of the incoming call: • from a stationary phone, • from a mobile phone • from abroad Function Tariff to ANI is used to change tariff for client depending on where he is calling. In order to use this function effectively, it is necessary to transfer from the public network the correct Caller ID of a client who makes the connection.

Adding prefixes Prefixes and tariffs assigned to them can be displayed by clicking Other->Tariff to ANI node.

In order to configure the service 'tariff to ANI' correctly, one has to follow the steps below:

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In the program VSM or VSC one has to define the initial digits which show from where a client is calling. For instance in Poland the callers who make a phone call from a mobile phone have their Caller ID starting with 60 or 50. Other numbers, starting with a different prefix, indicate stationary phones. Within the frames of this example, one should define the prefix 60 in 'Tariff to Ani' and one ought to assign there to the new tariff which will become effective only if the initial digits of the Caller ID are 60. • If the Caller ID starts with 50, the tariff 'Device tariff' will be used. • If the Caller ID starts with 60, the tariff 'TestTariffHigh' will be used. • In case of another Caller ID, the tariff assigned to the client will be used.

The phone calls coming to VoipSwitch from the telephone network are made by authorizing the telephone operator's gate. In VSM or VSC such clients are defined as 'GW clients'. In order to use the function 'Tariff to ANI' one must tick such a client in the 'Tariff to Ani' checkbox.

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Having saved changes for such a client, it is possible to make connections and the appropriate tariff will be used according to the Caller ID. Example...

In the following example you can see, that one IVR Client (login 888) called three times number 333, but in every case with other Caller ID. For every call according to caller ID (1000, 620 and Testowe_1212) appropriate tariff is chosen as shown below:

Reported problems Clients calling VoipSwitch can use different gateways. Some of them are sending different Caller ID format. One operator sends the Caller ID in the above-mentioned format, and the other sends the Caller ID with the country code (48) at the beginning, hence the format is 4860. In

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order to avoid making two separate entries in the table 'Tariff to ANI' it is possible to modify Caller ID coming to VoipSwitch. Rules on how to make it are described here.

ANI Tariff used by resellers. Resellers cannot define their own tariff to ANI rules, but may use the ones defined globally. To do so every client added has to have "Tariff to ANI" option marked (see picture below). Enter to the Edit client page, mark "Choose tariff according to" option, then click Rules button. In the new window mark Tariff to ANI option and click OK.

An exemplary calls report is shown below:

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IVR Client called 3 times to VoipSwitch with different Caller ID and tariff is chosen appropriately to defined Tariff to ANI table.

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Tariff to DNIS This page last changed on Feb 24, 2008 by admin.

Tariff to DNIS

This function makes it possible to change the tariff which is used to calculate the cost for a client according to the dialled target number. Consequently, this enables to calculate the cost for the "calling cards" clients in various ways. It is enough to define just the initial digits of the dialled number and to assign the new tariff to them.

The prefix and the tariff are defined in the 'Tariff to DNIS' part of VSM or VSC.

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The picture above shows how two entries have been defined. Each entry will work for a different number dialled by the client. One number is free of charge for the client, namely the number 800, which in many countries is toll-free. The other one is a toll number (700). In the first case the client will be charged at a higher rate in order to cover the cost of the toll-free connection from VoipSwitch. This function may be applied for every type of clients. Each client has a checkbox in his definition, under Funds & Tariff tab as shown below:

After ticking that checkbox and confirming the change, the 'Tariff to DNIS' service will be activated for the selected client. Click here for example...

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Those are example calls made by this client, notice that tariff is different for those 2 calls:

In this example two calls were made to VoipSwich, the first call was made to the toll-free access number and "tariff toll-free 800" was used. Second call was made to normal access number, and "normal tariff 700" was used.

With Tariff to DNIS it's good to have normal tariff assigned to every client, so if one calls normal access number to VoipSwitch system no tariff change is made, but when a client calls to special access number (for example toll-free access number 800) tariff to DNIS option intercepts this attempt and changes tariff so a client pays more (because VoipSwitch carrier has to pay for toll-free connection)

Tariff to DNIS used by resellers Although resellers cannot define "Tariff to DNIS" rules, they may enable this option for their clients. In such case tariff choice is based on global "Tariff to DNIS" options. This rule is similar as Tariff to ANI used by resellers

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6.0 Browsing calls, reports, statistics, payments This page last changed on Feb 24, 2008 by admin.

Calls connected through VoipSwitch or failed are available to browse in VSC and VSM application. Any call can be traced using date, client, calling number and other details. It is possible to see summary reports with charts about gateways usage, popular directions and many others. Statistics can help to choose best destinations and modify dialing plan according to such knowledge. Browsing payments made by client help solve misunderstandings. This section describes every such feature and is divided into subsections listed below. • Calls • ° Time shift ° Export ° Calls clients ° Resellers calls • Failed calls • Reports • ° Export • Statistics • Payments

Calls History of calls connected successfully through VoipSwitch is available in this part. It is possible to filter calls by: • Date and time - it is possible to see calls made only in given time. There are helper periods available as Today, Last week, which allow to define exact date interval. • Called number - destination number dialed by a client. • Caller ip - ip number from which call was received, starting characters can be used for filtering. • Caller id - caller id from which call was made, starting characters can be used for filtering. • Duration - duration of a call, it can be defined also operator of comparison like or = defined value. • Cost - similar to duration but revenue calculated for a client is used. • Tariff - tariff used to calculate revenue for clients. • PDD - time of connecting to given number, the less time the better. • Route - destination used to terminate calls. Working with filters, saving them, sorting columns, context menu is described in section Common UI elements

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Fig. 1 Main calls window with simple filter It is possible to use a few criteria to see only specific calls. After clicking Apply button the list of calls will be refreshed with filtering conditions. Below grid with calls summaries are calculated for the filtered set of calls. Informations available in summary section are: • • • • •

Total revenue - summary of revenue calculated for clients. Avarage revenue - average revenue for all filtered calls. Total duration - total duration for given set of records. Average duration - average duration for calls. Total cost - summary of cost calculated by tariff assigned to destinations, if tariff is not set at cost 0. • Average cost • Total profit - summary of profit which is total revenue - total cost value. • Average profit - average profit for every call.

Time shift

This new option allows to show result using time shift. This value is set with hours and allows to modify call_start and call_end columns by adding hour value set there. It is useful when you want to compare calls being made to VoipSwitch with the same calls received from carrier and they are not in the same time zone.

Fig. 2 Using time shift option Export

Calls displayed in the list can be exported to CSV file. It is coma delimited and can be easily opened in Excel application or in Notepad. You may choose columns to export if it suits your needs. After choosing export file the name dialog (as shown on Fig.3) is shown with default export columns selected. Choose

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which column to export and click OK button.

Fig. 3 Customizing calls export Calls clients

Calls node can be expanded to show calls made by specific type of clients. Only by using specific type of client it is possible to filter calls for a specific client. Client login name must be put exact; it cannot be set as starting characters because it won't show any records. Other filtering fileds are the same.

Resellers calls

Special part of calls is availalble for resellers. It is possible to see calls made by every reseller on every level. Filtering by date is also available (as shown on Fig.4).

Fig. 4 Resellers calls

Failed calls It is possible to filter failed calls by: • Date and time - it is possible to see calls made only in given time. There are helper periods available as Today, Last week, or from which allows to define exact date interval. • Called number - destination number dialed by a client. • Caller ip - ip number from which call was received, starting characters can be used for filtering. • Caller id - caller id from which call was made, starting characters can be used for filtering. • IE Error • Reason -

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• PDD - time of connecting to given number, the less time the better. • Route - destination used to terminate calls. Working with filters, saving them, sorting columns, context menu is described in section 1.0 Common UI elements

Fig. 5 Failed calls with simple filter options Clicking on every row of failed calls loads the form below the list. Values there are similar as in the list but for IE error and reason there are explanation of errors which can be useful for finding error on destination gateway. In the form below additional fields are: • • • • •

Client login Client type Route type IE Error description Release complete reason -

Fig. 6 Failed calls detailed form

Reports This section is used to see reports for calls made. It is possible to group and filter depending on different criteria. For grouped record it can calculate: • • • •

Sum - calculate revenue for clients. Profit - profit as difference between cost of termination and revenue paid by client or reseller. Average Count

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•

• • • • •

Fig. 7 Reports window with simple example. Grouping allows to see sums for chosen client, period of time and other options. Below is a list of possible grouping: Clients type - when client type is choosen it will show record grouped by logins from the chosen type. Using such report can be helpful to see clients calling best. To see all clients informations specific client login cannot be chosen. Route - when only type is chosen the system will show how traffic is divided between different gateways. Period - hour, day, month grouping is allowed. Resellers of any level - when reseller of any level is chosen the system will show how traffic is divided for every one of them. Country, region - using this grouping we can see best countries and regions chosen by clients. Visualization of this report is available in pie chart. Prefix - in comparison to country or region we can group by specific prefix, for example the same region description can have many prefixes so this option will give us more detailed grouping.

Defined criteria can be used as one or as many. Below please find list example reports: Period:Monthly + Route:External gateways - it will show how in every month for chosen dates traffic was sent between destinations defined in Gateways. Period:Daily + Group by:By country - it will show daily calling to different countries and regions. Period:Monthly + Group by:By country + Resellers:Res 1 - it will show monthly calling to different countries and regions for every reseller. Choosing many groupings can result in high CPU usage and long operation. It is better to limit time for which such report is being generated and use grouping with caution. Grouping records will be applied for records filtered. Available filters are: • • • • • • •

date interval resellers level specific reseller of chosen level client type specific client for chosen type route type specific route for chosen type

For some reports the charts generation is available. Grouping by country makes it possible to see as pie chart countries chosen by clients. For every reseller level it can be presented on a pie chart how big the usage of every reseller in total

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traffic is.

Export

Every generated report can be exported to CSV file format. This format can be opened using Excel. Only visible fields will be exported to file.

Statistics This section is using calls made and failed to calculate statistics. They are useful to check quality of gateways. Values available are: • • • •

ASR Number of calls made, calls failed Average, total, shortest and median of duration Best, worst, average, median PDD

Statistics are calculated depending on defined filters: • • • • • •

*Date and time period * - except for days it is possible to define period of hours. GW clients - it can be checked for this type of clients ( mostly used in wholesale ). GK/Registrar clients Specific client - it can be chosen by a client from one of types defined above. External gateway External gatekeeper -

Payments Section used to browse payments made by clients. Payments can be checked for filtered date, type of client, login and other criterias.

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7.0 Settings This page last changed on Feb 11, 2008 by rashid.

• Introduction • Types of settings • ° VoipSwitch ° - Call settings - Authorization - Other - Rerouting calls - Ending calls - Save failed calls when ° H323 ° - H323 listeners - Other ports - Gatekeeper - Authorization ° SIP ° - SIP listeners - Registrar - Other ° PC2Phone ° - Pc2Phone listeners ° Callback ° - Callback listeners - SMS Callback listeners - Regular Callback ° Callshop ° - Callshop listener - Callshop's web page addres ° VoipBox ° - Listener - Other settings - Time multipliers ° Invoices settings ° - Mail settings - SMTP settings

Introduction Settings are used to define parameters for VoipSwitch main system and for few modules. After clicking Settings node in VSM application it will expand and show different sections as shown on figure. !Settings.jpg! After changing most settings, VoipSwitch should be restared or at least start command Reload VSM data.

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Settings defined there are used mosty by VoipSwitch, every change there has influence on VoipSwitch behaviour and efficiency. Any change made here should be done carefully and with full awareness of possible effects. If not sure ask VoipSwitch support. Parameters here are divided into 5 sections:

Call settings

• Limit ring time - value given in seconds allows to define when call will be abandoned in case no one answers. It is useful to set some value there to avoid endless calling because then calls redirection will not happen. • Limit call duration - sometimes it can be useful to limit maximum call duration - value in minutes of longest possible call. • Use media timeout - special timeout used to disconnect both sides of conversation when media packets are not coming from one side during this time. It will work only in full proxy mode when media packets are coming through VoipSwitch. • Limit number of hops(re-routing policy) - it can be limited how many hops should be tried before ending call. Normally it is unlimited. If there are matching prefixes defined in dialing plan, all of them will be tried. This parameter allows to limit it. • Guest account - account used to authorize calls normally is not authorized to any client. Without this option set with login name all these calls would be rejected. When it is set calls will be authorized and billed for this client. Any option changed before it will be used by VoipSwitch must be saved and settings must be reloaded. It can be done by clicking right mouse button on Calls window of VoipSwitch and choosing from context menu command Reload VPSConfig data. After executing this command new settings should be applied to working VoipSwitch.

Authorization

Other

Parameters set in this section turn on and off functions available with VoipSwitch. If any of them is not used it should be unchecked and it can improve VoipSwitch effciency.

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• Use common clients - when unchecked common clients will not be able to call through VoipSwitch. • Save active calls in DB - when checked it will save information about connected calls to special database table. It is presented through VSC pages or limited version for resellers in VSR. • Use load balancing - when checked load balancing is available in dialing plan. • Use resellers - if this is unchecked VoipSwitch will not calculate any cost for reseller of any level. It will also not substract any value from their account. It can cause some problems if unchecked by mistake because all calls made by resellers clients won't be assigned to resellers. • Use time spans in DP - when checked it is possible to define time span for dialing plan positions.

Rerouting calls

Rerouting calls allows to find different gateways for dialed number if for any reason a call cannot be connected. VoipSwitch is set by default to reroute all calls no matter what error code is received from termination gateway. Sometimes it can be required to stop rerouting calls so calls can be treated as failed faster then waiting for trying many gateways. Good example is error "User busy". This message is returned when dialed number is busy. There is no sense in finding other gateway because it will still be busy. Using this dialog can be an added option allowing not to reroute when such error will be received on VoipSwitch. Similar rules can be added for every error returned by gateway. Rules set here are applied to all numbers defined in dialing plan.

Ending calls

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For some errors specific for VoipSwitch it is possible to define error number sent to SIP and H323 clients. It is important to set it for both protocols for every reason. Possible reasons which could happen in VoipSwitch are: • Destination offline - it can happen when defined gateway is behind firewall or there is a mistake in IP address for such destination. • Number doesnt exists in dialing plan or tariff - it will occur when for dialed number a dialing plan entry or prefix in tariff couldn't be found. • Unauthorized call - VoipSwitch denies call processing because of unauthorized request, IP not added, wrong login and password sent. • Codec problem - codec used by a client doesn't match a list of available codecs in client definition. • Uknown reason • Channels limit - number of allowed channels for client or gateway exceeded. Second part of these settings is for changing error numbers received from destination and passed to clients. Default behaviour is to pass it unchanged but using these settings it is possible to replace one error number with another. There are two buttons which allow to chose error number. One is Gateway end reason and number set there will be replaced with End reason sent to client. After clicking button rule it will be copied to text box on the right and for next calls matching Gateway end reason it will be replaced. Adding error number replacing rules doesn't require restarting VoipSwitch. Save failed calls when

Calls failing because of VoipSwitch misconfiguration are not stored by default in database. It was built in this way to avoid counting such calls for general statistics like for example ASR. Sometimes in order to have more extended information about such calls it is possible to turn on saving them in failed calls.

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Every error presented on screen above can be checked to be saved in database in general failed calls table.

H323 This section deals with setting H323 listeners, ports, gatekeeper and authorization of H323 devices.

H323 listeners

Listeners are needed for clients to connect to server. Server is listening on it's IP (it may be one or more IP addresses, but according to our experience no more than 5) for clients to connect. You may setup H323 listeners in section shown on Fig.1. Left box (Available computer addresses) is showing all IP addresses assigned to server network adapters. If you wish to choose one of them to be used for clients to connect just select one and add to the second box by clicking ">" button between boxes. There is one IP address (79.187.62.139) chosen on example Fig.1. Every VoipSwitch executeble file is prepared to operate on fixed IP addresses list. If a server has more than one IP one should make sure that VoipSwitch is able to work with all of server's IP addresses. VoipSwitch needs global IP address to function properly. It is impossible to use VoipSwitch on Private IP addresses. H323 listener has default port 1720. If you wish to add more ports just choose one and click green "+" (plus) button. Newly added port will be listed above (H323 listener ports list shown on Fig. 1). If VoipSwitch is unable to start one or more listeners check if some application is not using this port already. Click here to read more

Fig.1 H323 listeners section If you add more H323 listener ports, VoipSwitch will listen those ports on every choosen IP address. After changing listeners IP addresses or ports you have to restart VoipSwitch or at least listeners. Other ports

This section has only two fields. You may choose starting UDP ports of media and gatekeeper's RAS. Default settings are 6000 for UDP media and 1810 for GK's RAS and shouldn't be changed without a reason.

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Fig.2 H323 other ports section Gatekeeper

This part of H323 settings allows you to assign IP and port for VoipSwitch Gatekeeper. Choosing IP address and port is similar to H323 listeners. Default gatekeeper port is 1719.

Fig.3 H323 gatekeeper section Authorization

Authorization section allows to turn on or off user login option by H323 ID. H323 login consists of login, password and separator. By default separator is at (@) sign and H323 login looks like: username@password

Fig.3 H323 authorization section H323 user and password separator may be changed in this section. Default settings are shown on Fig.3.

SIP Whole SIP Settings window is shown on this picture (click to view).

SIP listeners

Listeners are needed for clients to connect to server. Server is listening on it's IP (it may be one or more IP addresses, but according to our experience not more than 5) for clients to connect. You may setup SIP listeners in section shown on fig. below.

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Similar to other listeners (H323, PC2Phone, etc.) consider that every VoipSwitch executable file is prepared to operate on fixed IP addresses list (or single IP - depending on server configuration). Make sure that all server IPs are recorded in our CRM to avoid VoipSwitch failures. SIP listener has default port 5060. If you wish to add more ports just choose one and click green "+" (plus) button. Newly added port will be listed above (SIP listener ports list shown on previous fig.). If VoipSwitch is unable to start one or more listeners, check if some application is not using this port already. Click here to read more

Registrar

This part of SIP settings allows you to assign IP and port for VoipSwitch Registrar. Choosing IP address and port is similar to SIP listeners. Default registrar port is 5060.

Other

In this section you can setup Realm and User-agent for SIP protocol. Default settings (empty) are shown below:

In this situation Realm will be "VoipSwitch" and user-agent will be "VoipSwitch 2.0"

PC2Phone PC2Phone settings are limited only to listener configuration.

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Pc2Phone listeners

Under Settings/PC2Phone Settings Listener is needed for PC2Phone clients to connect to server. Server is listening on it's IP (it may be one or more IP addresses, but according to our experience not more than 5) for clients to connect. You may setup PC2Phone listeners in section shown on fig. below.

PC2Phone listener has default port 1800. If you wish to add more ports just choose one and click green "+" (plus) button. Newly added port will be listed at "PC2Phone listener" section.

Callback Whole Callback Settings window is shown on this picture.

Callback listeners

Callback listeners are needed for Callback function to work. Server is listening on it's IP (it may be one or more IP addresses, but according to our experience not more than 5) for Callback trigger informations.

Similar to other listeners (H323, PC2Phone, etc.) consider that every VoipSwitch executeble file is prepared to operate on fixed IP addresses list (or single IP - depends on server configuration). Make sure that all server IPs are recorded in our CRM to avoid VoipSwitch failures. Callback listener has default port 1801. If you wish to add more ports just choose one and click green "+" (plus) button. Newly added port will be listed above (SIP listener ports list shown on previous fig.).

SMS Callback listeners

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SMS Callback listeners are needed for SMS Callback function to work.

SMS Callback listener has default port 1802. If you wish to add more ports just choose one and click green "+" (plus) button. Newly added port will be listed above (SIP listener ports list shown on previous fig.). Additional feature to set is Charge calls only when both legs were connected. If set failed callback calls will not be charged at all. When VoipSwitch owner has to pay for callback call this may cause money loss and has to be used with care.

Regular Callback

In the section Callback one sets parameters of Callback listeners. Default values should work fine, so do not change this settings without a reason. Regular Callback Settings should be configured to fit one's needs.

Regular Callback settings node in VoipSwitch Manager. In this program there are 4 groups of settings:

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• • • •

Common, ANI Callback, PIN callback DID callback All parameters set here are subsequently used by a twin program which functions as a service in the Windows system and is called "Calls Reader Service". One should remember that after every change the service "Calls Reader Service" should be re-started (in order to download new settings). The purpose of the section Common is to determine the settings used by the Service "Calls Reader Service" to connect to VoipSwitch. One should set the IP number of a server where VoipSwitch functions and the port are used as SMS Callback (by default it is port 1802). The parameter "Checking interval" determines the frequency - how many seconds the service "Calls Reader Service" checks for new connections with the number which activates callback. If this parameter is too big, the period of waiting for calling back will be to long; if this parameter is too small, the user may not have enough time to answer the call or the system may work too slowly. From experience it is known that the optimum time is 3-5 seconds. After finding out in the database table "CallsFailed" about an attempt to connect with the number which activates callback, the Service "Calls Reader Service" collects "Caller ID" and sends the information to VoipSwitch in order to try and execute a connection. The time after which this information is sent is known as "Sending delay". At the initial phase of using this service, one may select the option "Make logs". This option enables recording additional information about the functioning of the service "Calls Reader Service" in the Windows system logs. Later this option may be switched off. The purpose of the section ANI Callback is to define the number which activates callback and the numbers which the "Caller ID" will be connected with. It is possible to define one or several numbers. (It is not recommended that more than 5 numbers should be defined as this may slow down the functioning of the system). If more numbers are defined which activate callback and more than one number in the table 'IVR numbers', the appropriate number of the table 'Numbers' will be connected with a number in the same position in the table 'IVR numbers'. A part of ANI Callback is used for the clients who activate callback only from the numbers authorised earlier. See register option for more information about saving authorised numbers. Section PIN Callback is used during the execution of callback connections when "Caller ID" is not recognised. This option may be switched on and off with the check box "Check PIN if ani couldn't be found". Now it is necessary to verify a user who will be used to execute return connections, in the section "Client callback". It is necessary to select a user in the section PIN Callback from the list "Client callback", but the user will not bear any costs unless the PIN authorisation is used. Defining the numbers and the IVR numbers assigned to them is similar to such defining in the case of ANI Callback. One must remember that a number defined in the table 'IVR numbers' should indicate the PIN action of the system IVR.

Section DID Callback. This type of callback allows this service to work even when the "Caller ID" is not transferred to the system VoipSwitch. In this type, each client is assigned a unique number to which the client calls. The call is not connected as in the case of other callback calls but the system, on the basis of this target number, calls back to a previously defined number assigned to this client. All target numbers

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defined for clients must start with a common prefix (see Fig. 4, number 2000 in the "Dialing plan prefix"). In the same location, one defines the number the clients will be connected with (number 777 entered in the dialing plan). This is a number for the IVR action of the system VoipSwitch. This is an action of the type "Ask for number" because the clients do not require authorisation. Calling a specific target number triggers the connection of a particular client number with the number assigned to the IVR action. There is no need to call from specified number (as before), so Client is able to trigger callback call from any phone device, the system will call back to specified number assigned to this Client . When adding DID callback prefix use number that is not used in any of existing dialing plans or add dedicated dialing plan entry for this prefix and route to offline gateway as before Assigning the activating numbers and the numbers to which the callback will be executed for individual clients may be done in the window which appears after pressing the button "Clients DID numbers".

DID numbers window. In the left part of this window, there are the already assigned activating numbers together with relevant users and the numbers which will be used to connect with a client. In the above image, in the first line there is information that if a call to the number 2000 1 is executed and is not connected, then the system should call back as the callback client of the login "800123" to the number 800123 and then connect this call with the number 777 defined in the previous window. The numbers DID, presented here, constitute only the last part of the activating number (in this example just "1" for number "20001").

Callshop

Callshop listener

Default port used for callshop is 1804. Using this part of settings it can be set IP on which listener for callshop will be started and also this default port can be changed to some other value. It cannot be used for port number already occupied by other application. After changing port or IP of callshop listener VoipSwitch application must be restarted. This change must be reflected also in callshop.ini file located in callshop working directory. In NETWORK section of this file parameter SERVER_PORT=1804 should be

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modified.

Callshop's web page addres

Valid address of Web or Portal module should be set. This link will be used to show calls history and other details about callshop client within callshop application. It opens automatically when clicked on Web tab in callshop application. Format of such link is: http://server.ip/Portal or http://server.ip/Web/

server.ip should be replaced with the IP address or domain name of the server that has the module installed.

VoipBox Any parameter used by Voipbox can be set in VSM application after clicking node VoipBox in Settings part. Parameters defined here are used by VoipBox and in general are taken only when it is starting. Any change made to them requires restarting VoipBox application. Exception of these parameters are time multipliers where defined parameters can be changed without restarting VoipBox.

Listener

Use VoipBox - determines if VoipBox is used or not. VoipBox IP Address and Port - enter here a valid IP number and port number of computer running VoipBox application.

Other settings

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Maximum number length - Determines limit of digits in dialed number. Non activity timeout - Determines the time in seconds, after which the VoipBox will send the dtmf selected by a user to the VoipSwitch system. If nothing has been entered, the system will repeat its request to enter the pin number or the telephone number we wish to be connected to. Finish key - A person making a call into the IVR system is asked to enter the pin number and then the telephone number which he or she wishes to be connected to. After entering the pin number or the telephone number, one can press the symbol defined as the finish key in order to make the verification or connection to the desired number faster. Having received that symbol, the VoipBox will automatically start analysing the entered number. When the user does not press this symbol, the analysing of the entered pin number or telephone number will start only after the period described as "non activity timeout". Redial string - It is used to redial the last dialed number. Working for IVR, callback clients. It is working only for numbers dialed during current session. First number dialed after being connected must be picked up manually or using speed-dial. End call string - A user may terminate the connection at any moment. In order to do so, a user has to press the symbols described here. The telephone call will become instantly disconnected and the system will ask the user to enter another telephone number. Non activity retries - This value defines how many times the system will ask a user to enter the pin number or telephone number. The request will be repeated according to the defined value only if a user does not enter any digit.

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Wrong pin retries - Defines how many times the system will repeat the request to enter the pin number in the event that an incorrect pin has been entered. If the user fails to enter the correct pin number, he or she will be disconnected. Time multiplier - This multiplier may be used to change the message about the remaining time of the telephone call, which is communicated to the user. Only the message will become changed as the actual time depends on a specific tariff and is unchangeable. If we set the multiplier at 1.1, the message will communicate that the remaining time is increased by o 10% compared to the actual time after which the telephone call will become disconnected. Time addition - With this variable, it is possible to change the information concerning the remaining time of the telephone call by a certain number of seconds. If this value equals 10, the user will hear that the remaining time of the telephone call is 10 seconds longer than the actual time. Round time to minutes - Thanks to this option the information concerning the remaining time of the telephone call will become rounded off to the nearest whole minute and the number of seconds will not be included in the message. Silence duration - This parameter is defined in seconds. Each message becomes sent (ask for pin, ask for number, account information) to the client only after this period of time. Use client's account to recharge - If the setting Use client's accounts to recharge is set to on, the account can be recharged using the IVR or common client password. In such case the amount of credit that is present on the account of the users whose password one uses as the PIN number is added to the account of the user who makes the recharge. The account of the customer that is used to make the recharge is zeroed, and a Return type payment containing the credentials of the user that made the payment is added. Import dialable scenarios - This button allows to import saved dialable scenarios. Usually scenarios are located in subfolder \scenarios\dialable in VoipBox directory. Select language pair - In this field you should enter language names and digits assigned to them, for example: 1-English;2-Spanish. More information in section Save changes - Any saved changes in VoipBox configuration require restarting VoipBox.exe in order to activate them.

Time multipliers

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This part of settings is used to define multipliers used for resellers or lots. Values set there are used to modify time announced to clients using IVR services. Value set for lot will apply to all clients belonging to it and value set for reseller will be used for all clients created by reseller. Value of this parameter is used to multiply rate for number dialed by client. It is used only to change time being announced.

Invoices settings In order to modify invoice settings choose the "Settings" menu and next "Invoices". The invoice settings window will appear (Fig.1).

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Figure 1. Invoices settings in VoipSwitchManager You subsequently enter: The directory in which the documents will be saved - "Output folder" (Fig.2).

Figure 2. Setting output folder for generated invoices. You can choose the directory by pressing the button at the right side of a field (Fig.3). A window will appear, in which you select the appropriate directory and press "OK". Then we enter a seller's name, address and Tax Identification Number which will be printed in the invoice (Fig.3 ).

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Figure 3. Setting seller details. All bolded items are required for generating invoices. The following settings concern the invoice (Fig. 4). They are "Terms of payment", "Invoice item", "VAT, PST rate", "Currency symbol", "Decimal places", "Invoice number" and "Place of making out".

Figure 4. Invoice detailed settings. In the name of an invoice item you can use variables which will be changed into appropriate values when generating an invoice. You can use the following variables: • • • • •

[FROM_DATE] - the date since which the billing is generated. [TO_DATE] - the date until which the billing is generated. [LOGIN] - the login of a client for whom the billing is generated. [CLIENT_NR] - the number of a client for whom the billing is generated. [ACCOUNT_STATE] - the account state of a client for whom the billing is generated. Example The value "Calls from [FROM_DATE] to [TO_DATE]" when generating an invoice for the period from 2005-12-01 to 2005-12-31 will be changed into the value "Calls from 2005-12-01 to 2005-12-31". Variables can be edited manually but you can also use the button on the right side of the field which brings out the list of variables and their description. In order to enter the appropriate variable you can choose it from the list and press the "OK" button. It will be added at the end of the invoice item name.

Next you have to quote the VAT rate and a currency for the invoice. You can also specify PST rate and number of decimal places for amount. The next step includes defining the way of invoice numeration. In the invoice number you can use the following variables:

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• [YEAR] - the year of issuing an invoice. • [MONTH] - the month of issuing an invoice. • [NUMBER] - the following invoice number. Example The value "INV/[NUMBER] with the number 20 will be changed into the value "INV/20" when generating an invoice. Variables can be entered manually, you can also use the button on the right side of a field, which brings out the list of variables and their description. Then we specify the way of generating the following invoice number (the "[NUMBER]" variable).

Figure 5. Resetting invoice numbers. If the number should be reset you have to select "Reset invoice number" and specify if the numeration should be started from the number 1 every month or every year. You also have to specify what invoice template you want to use (Fig. 6).

Figure 6. Setting invoice template and footer text. There are two templates available - the standard one and the template which does not include the VAT tax. After pressing the "Edit templates" button you may choose which template you wish to edit and Template Designer will open. There you will be able to edit template. Template editing is only for advanced users. Use this feature with care! Then we specify the invoice footer. In the footer (Fig. 6) you can use the same variables like in the invoice field in the same way. If you want your logo to be printed in the invoice press the "Select logo file" (Fig. 6) button. It brings up the window in which we point an appropriate graphic file (jpg, gif, bmp). The section "Summary billing grouping type" (Fig. 7) specifies the way of generating the summary billing. If you select "Tariff prefix and description" the calls are grouped according to the prefix and tariff description, if you select "Tariff description" the calls are grouped only according to tariff description. The section "Additional grouping type" (Fig. 7) specifies whether additional monthly, daily grouping should also be created for the summary billing or not.

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Figure 7. Billing grouping and summary generating options. Send invoice" means that an invoice (and the billing if it is also generated) should be sent to a client by e-mail. "Create detailed billing" means that a detailed billing (including all calls) should be generated for a client. "Create summary billing" means that summary billing should be generated for a client.

Mail settings

In order to modify settings of sent e-mails choose the "Mail settings" sub-menu from Invoices. It brings out the settings window for e-mails sent to clients together with invoices.

Figure 8. Invoices mail settings. You have to enter as follows: the address of an e-mail from which you will send messages, title and content of a message. In the title and content of a message you can use the following variables: • [FROM_DATE] - the date from which the billing is generated. • [TO_DATE] - the date until which the billing is generated. • [INVOICE] - the number of an invoice. Variables can be edited manually but you can also use the button on the right side of the appropriate fields which brings out the list of available variables.

SMTP settings

The parameters of connection with the SMTP server should be set in the section called "SMTP settings".

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Figure 9. SMTP settings for sending invoices. You have to enter as follows: SMTP server address, if a server requires authorization you have to select "Server requires authorization" and enter a user's name and a password. After entering the parameters you can press the "Test settings" button in order to test your settings. A window will appear where you can enter the address to send a test e-mail. If the settings are incorrect a message window will appear with description of an error. Otherwise the message will appear like on Fig.10.

Figure 10. SMTP settings test success.

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8.0 Services This page last changed on Feb 21, 2008 by admin.

VoipSwitch ® Services give possibility to automate some cyclic tasks. This section of documentation covers configuratin and how-to-use examples for VoipSwitch ®Services. • Configuration • ° Starting and stopping VoipSwitch Service ° SMTP settings • Services log • Account state • Account state reseller • Expiration time • Archives • Invoices • Payments • Voice Messages

Configuration Starting and stopping VoipSwitch Service In order to start and stop the service, select accordingly Start or Stop position from the menu which appears after clicking the Configuration leaf in Services section of VSM.

Fig.1 Starting and stopping VoipSwitch Service The buttons "Start" and "Stop" are used to start and stop the service. Label above buttons indicates the current state of services. It may be either RUNNING (as shown on Fig. 1) or STOPPED.

SMTP settings Next, you should provide the SMTP server parameters, i.e. server address; if the server requires authorisation, check the box "Server requires authorization" and enter the user name and the user password. Having done that, if you want to test the parameters, you may press the button "Test settings". You will see a window where you enter the e-mail address to which you want to send a test e-mail message.

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Fig.2 SMTP settings If the settings are incorrect, the message with the error description will appear (see Fig.3a), otherwise, the message will look as shown on Fig.3b. a.

b.

Fig.3 SMTP settings test result. a) failed, b) succeed Having tested the connection parameters and SMTP parameters, save the settings by pressing the button "Save". SMTP settings are needed for sending emails by Account state, Invoices and Payments modules.

Services log This is the window with the log of the service operations.

Fig.4 VoipSwitch services log The upper part of the window contains the list of operation types and date fields. Select the starting and ending dates of a period and you will be able to see the operations executed within this period. Having changed the operation type or dates, press the button "Apply filter" in order to refresh the content of

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the list. You may arrange the list according to different criteria by clicking on the appropriate column heading.

Account state At the "Account state" leaf, you may define how often the system should examine the state of user accounts and send reminding messages by e-mail.

Fig.5 Account state controls - check interval. The minimum value of check interval is 30 minutes. The service examines whether a given type of e-mail has already been sent to a particular user and does not send the e-mail again until 7 days later. For example, if the user X account's state is EUR 1, and the e-mail message is defined for the state below EUR 2, regardless of the above setting, the e-mail message will be sent every 7 days unless the account state changes. Below, there are checkboxes to define whether the service should use the client types or " Lots ".

Fig.6 Account state controls - client types or lots. Then, from the selection lists, choose the client type you are interested in and (if "Use lots" above was checked) a relevant "Lots".

Fig.7 Account state controls - client type choosing. After making the selection in the list below, there will appear a list of messages defined for a client type (or "Lots").

Fig.8 Account state - list of messages defined for a clients. In order to add a new message with the information about the account state, press the button "Add". The message-defining window will appear.

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Fig.9 Account state - email message composer. First, enter the account state, at (or below) which the reminding message will be sent. Then, define the message title and the text of the message. In case of both the title and the text of the message, you can use the following variables: • [AMOUNT] - current state of the account • [LOGIN] - client's login • [NAME] - client's name The variables may be entered manually or by selecting particular positions from the list of all available variables in the drop-down menu. To open the drop-down menu (see Fig.10), press the 'upside-down triangle' button to the right of the variable field.

Fig.10 Account state - special variables window. After pressing the button "OK", the selected variable will be entered at the end of a relevant field. The variables will be replaced with appropriate values while the message is being sent. Example When sending a message to a client whose account state equals EUR 1, the message title "Account state: EUR [AMOUNT]" will be changed into "Account state: EUR 1". It is also possible to send HTML messages. To do so, check the box "Use HTML", and then select an external, previously defined HTML file (by pressing the button located to the right of the field "HTML file"). Having completed defining, confirm the new message by pressing "OK". In order to edit a previously defined message, highlight the relevant position in the list and press the button "Edit" or double click on this position in the list. In order to remove a message, highlight the message in the list and press "Delete". Remember to press the button "Save settings" after defining is completed (as shown on Fig.11).

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Fig.11 Account state - save settings.

Account state reseller Settings in this section are very similar to the previous one. Instead of "Client type", "Reseller level" selection is available.

Fig.12 Account state reseller controls.

Expiration time This service is used to block clients accounts after defined time which passed from first call. Common usage is to set expiration time on 1 month after first call so user must finish all his funds in a month. After this time even if his account is not 0 he will be blocked from calling. In the upper part of the tab "Expiration time", you may set the frequency of how often the service should examine the expiration time of user accounts. The minimum value of expiration is 1 hour. Second time which can be set there is time of checking interval. Minmum value of it is 30 min and it means only how often service is checking for expired clients. Setting this value higher can spare system resources.

Fig.13 Expiriation time. Below, there are checkboxes to define whether the service should use the clients, client types or "Lots" during such examination.

Fig.14 Expiriation time - use clients. If you choose "Client types", for each client type you must provide the time from the first call after which the user account will be blocked.

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Fig.15 Expiriation time - selecting all clients of particular type. If you do not want to have the accounts for some client types blocked, uncheck the box to the left of a relevant type. If you decide to use Lots , the lower part of window contains the list of "Lots" for which the expiration time will be examined.

Fig.16 Adding new lot to expiriation module. In order to add "lots" select the time from the last call (1.), after which the client account in a selected "Lots" will be blocked; next choose the client type and desired lot (2.) from "Lots" drop-down list. After defining is completed, press the button "Add new" (3.). After that new expiriation entry will be added to the list (Fig. 16, green line and elipse) In order to edit previously chosen Lots , highlight the appropriate position in the list; then change values and press the button "Save". In order to exclude "lots" from examination, highlight selected "lots" in the list and press the button "Delete". Account will expire cruelly, with aside of payments and client's account state.

Archives This service gives possibility to archive old data from calls and failed calls tables to reduce database tables size.

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Fig.17 Archives - main controls. Checked interval options define a number of days between each archive and exact execute time.

Fig.18 Archives - time interval settings. Move to archive records: 1. older than X months - on execution time the system analyzes calls tables and saves records older than X months to table named "callsarchive_Y", where Y is number starting from 1 2. when record counts exceed X records - after condition is true (number of records is more or equal X), the oldest records are transferred to archive Create new archival table with current active archive: 1. contains records from X months - when archive table contains data older than X months a new table is created and archive will be stored in the new table 2. records count exceeds Y records - when archive table contains more than Y records a new table is created and archive will be stored in the new table

Fig.19 Archive options.

Invoices In the upper part of the tab "Invoices", you should define how often the service should generate invoices. The minimum value is 30 minutes.

Fig.20 Invoices - time interval and range of invoice generation. In the next step, define the period for which the invoices are to be generated. Next, choose a destination folder into which the invoices will be generated.

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Fig.21 Invoices - select output folder. Choose the folder by clicking the button on the right and selecting the appropriate folder. Below, there are checkboxes to define whether the service should use the client types or Lots while generating invoices. If you select "Client types", you should define for what client types the invoices will be generated.

Fig.22 Invoices - selecting client types or lots. If you want the application to generate invoices for a certain client type, check the box to the left of the relevant type name. If you have chosen "Use lots", the invoices will be generated for the clients in the "Lots" of the below list.

Fig.23 Invoices - selecting all clients of particular type. In order to add "Lots", select a client type and next the "Lots" you are interested in and press the button "Add new". In order to remove a "Lots" form the list, highlight it in the list and press the button "Delete".

Payments Payments service is used to charge users accounts with desired amount. Payments are made cyclic on daily, weekly, or monthly basis. There is a posibility to send notification email to the client at some time before charge. Notification email is sent only when client account state is lower than scheduled payment. In order to enable sending notification emails corresponding checkbox must be marked. Reminder template may be edited. An example is shown below:

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Fig.24 Invoices - reminder template. You should define whether the service should use the client types, lots or single clients while executing payments.

Fig.25 Invoices - using client type, lots or list of choosen clients.. When you use client types, as shown above, mark client type you want to charge and set parameters up. The parameters are: 1. charge every - set time interval between charges; this may be some number of days, weeks or months. 2. fee - set the amount the client account will be charged. 3. start from - set start date of payments (useful i.e. when client has free of charge trial period). 4. payment desc - set description of this payment. Examples on Fig.25 Some examples of Payments are visible on Fig.25, especially how to setup weekly, daily and monthly payments to clients. When use lots is enabled, you have to set up charge settings for every lot you wish to charge clients from. An example is shown here:

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Fig.25 Invoices - use lots type. First choose one of client type (1.), next choose lot from Lots list(2.); then set up charge settings (charge interval, fee, start date and description) as in previous section (3.) and click Add new button(4.). Next, the new entry will appear on defined payments list (5.). Every defined payment is shown on the list (with short description and client type displayed) and may be edited or deleted. When use clients is enabled, almost all settings are similar to the previous section. Instead of lots the clients are added to list as shown below:

Fig.25 Invoices - use client types. Using this mode you can schedule more than one payment for Clients. For example Client can be charged monthly with given sum and also weekly with another sum. It is very usable when different Clients have different services.

Voice Messages Voice messages service sends an email to the configured user email address when they "miss a call". You may set the frequency of how often the service should check for new messages. Reminder message template may be edited. An example is shown below:

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Fig.26 Voice Messages - reminder template.

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9.0 Invoices This page last changed on Feb 15, 2008 by rashid.

• Invoices settings • ° Invoices settings ° - Mail settings - SMTP settings • Generating Invoices • Review of generated invoices. • Invoice example with description • ° Invoice main file ° Detailed Billing ° Summary Billing

Invoices settings Invoices settings In order to modify invoice settings choose the "Settings" menu and next "Invoices". The invoice settings window will appear (Fig.1).

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Figure 1. Invoices settings in VoipSwitchManager You subsequently enter: The directory in which the documents will be saved - "Output folder" (Fig.2).

Figure 2. Setting output folder for generated invoices. You can choose the directory by pressing the button at the right side of a field (Fig.3). A window will appear, in which you select the appropriate directory and press "OK". Then we enter a seller's name, address and Tax Identification Number which will be printed in the invoice (Fig.3 ).

Figure 3. Setting seller details. All bolded items are required for generating invoices. The following settings concern the invoice (Fig. 4). They are "Terms of payment", "Invoice item", "VAT, PST rate", "Currency symbol", "Decimal places", "Invoice number" and "Place of making out".

Figure 4. Invoice detailed settings. In the name of an invoice item you can use variables which will be changed into appropriate values when generating an invoice. You can use the following variables: • • • • •

[FROM_DATE] - the date since which the billing is generated. [TO_DATE] - the date until which the billing is generated. [LOGIN] - the login of a client for whom the billing is generated. [CLIENT_NR] - the number of a client for whom the billing is generated. [ACCOUNT_STATE] - the account state of a client for whom the billing is generated.

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Example The value "Calls from [FROM_DATE] to [TO_DATE]" when generating an invoice for the period from 2005-12-01 to 2005-12-31 will be changed into the value "Calls from 2005-12-01 to 2005-12-31". Variables can be edited manually but you can also use the button on the right side of the field which brings out the list of variables and their description. In order to enter the appropriate variable you can choose it from the list and press the "OK" button. It will be added at the end of the invoice item name. Next you have to quote the VAT rate and a currency for the invoice. You can also specify PST rate and number of decimal places for amount. The next step includes defining the way of invoice numeration. In the invoice number you can use the following variables: • [YEAR] - the year of issuing an invoice. • [MONTH] - the month of issuing an invoice. • [NUMBER] - the following invoice number. Example The value "INV/[NUMBER] with the number 20 will be changed into the value "INV/20" when generating an invoice. Variables can be entered manually, you can also use the button on the right side of a field, which brings out the list of variables and their description. Then we specify the way of generating the following invoice number (the "[NUMBER]" variable).

Figure 5. Resetting invoice numbers. If the number should be reset you have to select "Reset invoice number" and specify if the numeration should be started from the number 1 every month or every year. You also have to specify what invoice template you want to use (Fig. 6).

Figure 6. Setting invoice template and footer text. There are two templates available - the standard one and the template which does not include the VAT tax. After pressing the "Edit templates" button you may choose which template you wish to edit and Template Designer will open. There you will be able to edit template.

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Template editing is only for advanced users. Use this feature with care! Then we specify the invoice footer. In the footer (Fig. 6) you can use the same variables like in the invoice field in the same way. If you want your logo to be printed in the invoice press the "Select logo file" (Fig. 6) button. It brings up the window in which we point an appropriate graphic file (jpg, gif, bmp). The section "Summary billing grouping type" (Fig. 7) specifies the way of generating the summary billing. If you select "Tariff prefix and description" the calls are grouped according to the prefix and tariff description, if you select "Tariff description" the calls are grouped only according to tariff description. The section "Additional grouping type" (Fig. 7) specifies whether additional monthly, daily grouping should also be created for the summary billing or not.

Figure 7. Billing grouping and summary generating options. Send invoice" means that an invoice (and the billing if it is also generated) should be sent to a client by e-mail. "Create detailed billing" means that a detailed billing (including all calls) should be generated for a client. "Create summary billing" means that summary billing should be generated for a client.

Mail settings

In order to modify settings of sent e-mails choose the "Mail settings" sub-menu from Invoices. It brings out the settings window for e-mails sent to clients together with invoices.

Figure 8. Invoices mail settings. You have to enter as follows: the address of an e-mail from which you will send messages, title and content of a message. In the title and content of a message you can use the following variables: • [FROM_DATE] - the date from which the billing is generated. • [TO_DATE] - the date until which the billing is generated. • [INVOICE] - the number of an invoice.

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Variables can be edited manually but you can also use the button on the right side of the appropriate fields which brings out the list of available variables.

SMTP settings

The parameters of connection with the SMTP server should be set in the section called "SMTP settings".

Figure 9. SMTP settings for sending invoices. You have to enter as follows: SMTP server address, if a server requires authorization you have to select "Server requires authorization" and enter a user's name and a password. After entering the parameters you can press the "Test settings" button in order to test your settings. A window will appear where you can enter the address to send a test e-mail. If the settings are incorrect a message window will appear with description of an error. Otherwise the message will appear like on Fig.10.

Figure 10. SMTP settings test success.

Generating Invoices In order to generate new invoices from the „Invoices" menu you have to choose the "Generate invoices" item. There appears the window of choosing clients for whom invoices will be generated (Fig.1).

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Figure 1. Selecting clients to generate invoice for them At the upper part of the window you specify the period for which invoices are generated. On the left side there is the clients list of currently selected type. In the list placed at the top you can choose a type of a client. The field below is designed to filter the list, i.e. entering "s" in this field will filter the list leaving only the clients whose logins start with "s". In order to choose a client you have to click on a record's marker. In the middle part of the window (Fig.1) there are buttons designed to choose clients. Using the button „>" you can choose the selected clients, using the button ">>" you can add all clients to the list. If a client was chosen for the first time and no clients details are entered there will appear information about missing clients data and the window (Fig. 2) where you have to enter a client's data for an invoice is opened after pressing OK button.

Figure 2. Entering missing clients data

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The chosen clients are on the right side of the window (Fig.1). In order to modify a client's data double-click on record. Then the client's data edition window will appear. In order to delete a client from invoice just mark proper record and press the "Delete" button; confirmation window will appear. Choose "Yes" to delete, "No" to cancel operation. After choosing clients for whom we want to generate invoices we press the "OK" button. In the following window (Fig. 3) there is a list of prepared invoices.

Figure 3. Preview of invoices before generating • In order to send an invoice to a client by e-mail select the position „E-mail". • In order to generate detailed billing select "Detailed billing". • In order to generate summary billing mark "Summary billing". You can change the specified above parameters for all invoices by clicking the Options button and choosing the appropriate position from the menu (Fig. 4). At the bottom part of the preview window (Fig. 3) there is a summary of the value of all invoices. There is also the button enabling a removal of an invoice from the list "Delete".

Figure 4. Options for all prepared invoices You may also edit properties of prepared invoices (Fig. 5). In the invoice edition window you can change its data. In order to confirm changes you have to press "OK".

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Figure 5. Invoice editor window After preparing all invoices we start the process of generating by pressing the "Generate All" button. The window will appear with a progress bar showing the invoice generation progress and information about current job. After completing generating the program will open the invoices folder (defined in Settings/Invoice menu). Flash animation You can watch exemplary invoices generating as a Flash animation. Click here to view animation.

Review of generated invoices. In order to get the list of all invoices generated earlier you have to choose the item "All invoices" from the "Invoices" menu. A window will appear containing the list of all invoices issued before.

Figure 6. Generated invoice review This window is very similar to the invoice preview window which was described earlier. In this window you can generate an invoice again (the button "Generate invoice"), send an invoice by e-mail (the button "Send invoice"), open an invoice or billing (the button "Open...), delete an invoice (the button "Delete") or edit an invoice (the button "Edit").

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Invoice file is generated according to Invoices settings in VSM described before. On the Fig.7 and Fig.8 you can see exemplary invoice with short description of customizable fields.

Figure 7. Example of invoice with short description (upper part). Number

Where to configure

Description

1

Place of making out the invoice. This vaule is taken from "Place of making up" field in VSM Setting/Invoices section. Typically value: city where the company is located.

2

Invoice number. Generated for every invoice, should be unique. (required)

3

Sellers details (name, address, city, Tax ID. etc.) Configured in Settings/Invoices section of VSM. (required)

4

Buyer details (name, address, city, Tax ID. etc.) Configured in Client's "Personal data" tab for every client type.

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Figure 8. Example of invoice with short description (middle part). Number

Where to configure

Description

1

VAT Rate, configured in Settings/Invoices section of VSM. (required)

2

Product name or Invoice Item, configured in Settings/Invoices section of VSM. (required)

3

Currency symbol, configured in Settings/Invoices section of VSM.

4

Terms of payment, configured in Settings/Invoices section of VSM.

Detailed Billing If you choose to generate detailed billing additional PDF file will be generated. Exemplary detailed billing is shown on Fig. 9.

Figure 9. Example of detailed billing with description.

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In the upper part of the first detailed billing page chosen logo is displayed. Below logo you can see a client name and a date range of current billing. There are also summary values of total calls duration and total calls cost. Next item on page is a table with detailed billing information. Table columns are: • • • • • • •

Nr. (Sequential call number) Called number Call start Call end Call duration Call cost Tariff (tariff's entry description)

Summary Billing Similar as before, if you choose to generate and attach summary billing additional PDF file will be generated. Exemplary summary billing first page is shown on Fig. 10.

Figure 10. Example of summary billing with description. Header of the first page of the summary billing is similar to detailed billing first page. There is logo, client name and date range of current billing in upper part of first page. There are also summary values of total calls duration and total calls cost. In the table below you can see grouped values (according to chosen grouping options in VSM Settings/Invoices).

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Common UI elements This page last changed on Feb 24, 2008 by admin.

• Context menu • ° All client's modules (GW Clients, PC2Phone Clients, CallBack clients ...) ° Automatic Account Generation ° Dialing plan ° General calls ° Calls for client types ° Gateways & GK/Registrar Modules ° Resellers I, II, III ° All invoices & Invoices preview. • Filters • Stored views • Data paging • Automatic data refreshing • Export and Import data • ° Modules export formats ° - Client's export - import: - CallShop clients export - Calls module (general and calls for client types): - Calls for resellers module - Failed calls - Dialing plan - Tariffs - Reports • Menu tree • Users • Tariffs comparer • VSM, VSC, VSR - AutoUpdate • Search client's • Validate prefixes

Context menu Note: VSM is an executable program and its user interface is independent from the Web interface. Therefore, we were able to integrate Context Sensitive menu feature . What could Context menu do for me? Most of VSM modules have a Context menu for each element. The Context menu contains helpful options that envoke related commands. This feature accomplishes tasks and completes your job quickly. For example, You want to check Client's reseller definition. Without the Context menu You probably would search "Resellers I" module manuly - now it's just one click. Remember: Context menu is always accessible by positioning the mouse pointer on the desired element and right-clicking the mouse. Context menu modules

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Let's take a little tour and check modules' menu options.

All client's modules (GW Clients, PC2Phone Clients, CallBack clients ...) Figure 1.2.1 displays Context menu for these Clients:

• Copy to clipboard This command allows you to copy basic client's data (like account name, password, reseller ID, tariff, remaining funds, credit allowed ... - in other words - exactly one row of data from the list is copied - without payment's history, or IP/ANI numbers). Data is copied in CSV (Comma - spaced values) - so it's possible to paste it directly to office applications like Excel or Word. Of course, you can also paste client data in VSM - for example, suppose that you want to copy GW Client information to Common Client . Note If you also want to move payments and ip numbers please use Cut to clipboard command. For example, if data row is copied from VSM GW Client's and pasted to MS Word it will look's like this: Clients smallgw,smallgw,301989979,1,0.0000,,-1,-1, Row format is exactly the same as Export - Import format. Please see A 1.6 for detailed description of client's export fields in this format. • Cut to clipboard This command works similar to Copy to clipboard. You can Cut additional data like: payments, calls, failed calls, IP numbers, and ANI numbers . Of course, after Paste, the original record is removed. Cut to clipboard is great for changing client type. Rows Copied to clipboard format is the same as Copy to clipboard. Additional data isn't copied via clipboard, but is discovered during pasting by record identifier. • Paste from clipboard This option allows for pasting the Client's information in the Clipboard to the currently selected Client. This applies to the information copied to the Clipboard using Copy or Cut menu options. This option is enabled only when clipboard data has the expected format. First line has to contain: Clients. VSM will delete the original information if the Pasted information was copied to the clipboard using the Cut option. Using Paste from clipboard option you could convert between each client type - except Callshop

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clients. This is because of dissimilar client type roles. For experts As described above Cut & Copy format is the same as Export & Import record format. Therefore, if you get some records from exported file and add at first line: Clients you could paste only some of exported items! This can be really useful! • Show reseller If a current client has a parent reseller, this command allows you to open directly reseller's editor with the client's parent reseller loaded into it. • Show tariff rates This command is enabled when the currently selected client has attached a defined-in-system tariff. It shows client-tariff rates. You don't have to search tariff manually - one click and you've got all you need. • Show client's calls... This command opens Calls module and filters calls for the currently selected client. Default date filter is "Today" allows you to see where this client called today (00:00 - 23:59).

Automatic Account Generation See a Context menu for a client shown on figure 1.2.2:

Fig. 1.2.2a: LOT context menu 1 • Add to reseller ... This command adds current lot and all of it's clients to the selected reseller. You could for example generate 1000 clients and then assign them to reseller I by using Add to reseller option. Reseller can then manage these clients in his VSR manager. ADD to Reseller option is the simplest way to generate clients for your resellers.

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Fig. 1.2.2b: Select reseller dialog

Dialing plan

See a Context menu for a client shown on figure 1.2.3:

Fig. 1.2.3a: Dialing plan context menu • Show route This command displays route-editor loaded with the route currently assigned to the selected Dialing plan item. This command is enabled only when the assigned route exists in the system and is accessible. Depending on the route type, different editors will be opened - for example GK/Registrar client editor, PC2Phone client editor, gateway editor .. Figure 1.2.3b shows GK/Registrar client editor in action, for editing route.

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Fig. 1.2.3: Show route with GK/Registrar client editor loaded • Show route calls This command is used when you want to see calls completed using the route assigned to the selected dialing plan. The default date filter will be "Today" (00:00 - 23:59).

General calls

See a Context menu for calls shown on figure 1.2.4a:

Fig. 1.2.4a: General calls context menu • Show tariff rates... This command is enabled when a current call is made using existing-in-system tariff. It shows tariff rates for the current call's tariff. This allows you to check if the selected call's cost and tariff are correct by just one click! • Show call rate... This command is more detailed than "Show tariff rates..." because it shows directly the rate used for calculating the cost of a call. It uses tariff prefix to match rate, which means that sometimes there could be more than one rate matched. For example 4865, 48 ... Call rate(s) will be shown in context-hint (see Fig. 1.2.4b). If call tariff was deleted from the system - this command will be disabled.

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Calls for client types Figure 1.2.5 shows the context menu for Calls by Client type:

• Show tariff rates... Same as in General Calls (1.2.4). • Show call rate... Same as in General Calls (1.2.4). • Edit Client Opens client editor - depending on client type - and allows you to change client definition with one click.

Gateways & GK/Registrar Modules

Both modules have a Context menu as shown on figure 1.2.6:

Fig. 1.2.6: Context menu in GK & Gateways • Show routes... This command enables you to see Dialing plan items related to Gateway / GK/Registrar Client. When you select this command, the Dialing plan module opens and offers a list of Dialing Plan items routed to the Gateway or GK/Registerar client. This is very usefull when you want to delete all Dialing Plan items related to the Gateway or GK/Registrar you want to delete. Two clicks to do this!

Resellers I, II, III

All resellers modules have a menu shown in Fig. 1.2.7:

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Fig. 1.2.7: Resellers context menu. In Resellers I Show subresellers... will be disabled - because 1.st level reseller is the lowest Reseller level. In Resellers III Show reseller... will be disabled - because 3.th level reseller couldn't have any parent reseller. • Show Subresellers... This command shows the reseller on lower level (for example for Reseller III it will show Resellers II..), which is assigned to current reseller as a parent. Using this command, you can easily manage the resellers tree . • Show reseller... Opens parent reseller definition for the currently selected reseller.

All invoices & Invoices preview.

Figure 1.2.8 shows the Context menu in Invoices.:

Fig. 1.2.8: shows all Context menu invoices. This menu is available in Invoices preview and Invoice generation. However, in Invoice preview the Show invoice, Show detailed billing, and Show summary billing options are disabled, because in Invoices preview *.pdf documents haven't been generated yet. • Edit invoice Opens editor for currently selected invoice and allows changes made to a document before printing (or reprinting). • Generate invoice This command generates *.pdf document with invoice and/or detailed, summary billing. Also, you can use this command to regenerate invoices after deleting or moving specific invoices. • Send invoice This command sends the selected invoice to the Client's e-mail address. You can always change this address using the Edit invoice command. • Show invoice, Show detailed billing, Show summary billing These commands are active if the related *.pdf files exist - as shown on Fig. 1.2.8. You can generate the invoice to create the related *.pdf documents before you send it . After "Generate" - depending on invoice settings - you will be able to see the preview of generated billings and invoices.

Filters

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Filters are commonly used in all modules. They contain fields - connected with the currently shown information. It allows Users to find information or records they need. Fig. 2 shows a filter from Calls Module VSM, Fig. 3 shows the same filter but from VSC:

Fig. 2a: Calls filter from VSM.

Fig. 2b: shows Calls filter from VSC. You can simply specify filters for all fields and click Apply filter to see the result. For example, when you type "048" number in "Called number" filter field and click Apply filter, you will see all calls made to numbers that start with "048", for example "048653456023". You can also see only the calls made in the last week - simply select "Last week". Show all (VSM only) button will clear filter values to defaults. Please note that "Show All" shows all the records in Clients, Routes etc. but in Calls, Failed calls, Reports .. and other modules it maintains the filter data in effect. For example, it will show >>not all;',19,4,0,6,0,2400,0, Field meanings (countig from 1): 1. 2. 3. 4. 5. 6. 7. 8.

Prefix Priority Route type ID Route ID Tech Prefix Call type Dialing Plan type From day (0-6)

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9. 10. 11. 12.

To day (0-6) From hour (0000-2400) To hour (0000-2400) Balance share (0-100%)

Tariffs

Tariff rates are exported in CSV Format. Using this function You could easly exchange tariff rates between tariffs, or make backup for restore later. Tariffs CSV format: 106;Canada;23.0222;0;6;0;2400;15;1;1;0;0; Field meanings (countig from 1): 1. 2. 3. 4. 5. 6. 7. 8. 9. 10.

Prefix Description Voice rate From day (0-6) To day (0-6) From hour (0000-2400) To hour (0000-2400) Grace period Minimal time (in seconds) Resolution

Reports

In Reports, exact current view is exported. Current view columns set depends on filter values, so file format could change when You change filter. Filtering all columns, Your file format will be as follow: 2006-12-05;azeermis;7562;23.7579;128.1695;0.40267627;59;23.7579;23.7579; Field meanings (countig from 1): 1. 2. 3. 4. 5. 6. 7. 8. 9. 10.

Date Reseller Sum duration Revenue Revenue res. Avg. duration (minutes) Avg. cost Count Cost Profit reseller.

Menu tree

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User interface in both VSC and VSM is divided into separate modules. Each module is accessible from tree menu - shown on fig 7a and 7b. !1.0Menutree.jpg|align=center!

Fig. 7A: Modules tree menu in VSM

Fig. 7B: Modules tree menu in VSC

In VSM all tree-menu items are also accessible in main menu as shown on fig 7c.

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Fig. 7C: VSM main menu gives You access to all items from treeview. In some lower screen resolutions, to save screen work space, You could hide left tree menu in VSM and use only main menu. !1.0HideMainMenu.jpg|align=center! Fig. 7D: Show & Hide tree menu in VSM. When menu is hidden, You could navigate using main menu. Also here You have list of all currently opened windows!

Users VSM allows You to add more than one user to operate VSM. When at least one user is added, application will ask You to log-in during start, as shown on Fig 8a. Please don't understand VSM users as VoipSwitch users. This feature limits access to VSM settings only

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Fig. 8A: Log-in dialog.

Fig. 8B: When using users, at the top of main window user always see his login data, and could easly relogin as another user. Adding users, You could to set them permissions to use only selected commands from menu and windows. Doing this You have guarantee that user will not disturb any important settings, or see confidential data. For example You could add separate users to add GW Clients, another to set-up resellers etc. VSM stores data in Database, not in User-Folder (as other windows application that supports multi-user use) - so installs for different Windows-users doesn't resolve problems with permissions. Only Users module could do this for You. If You don't want to login to application on every startup, please remove all users. Then application will work in standard mode. Adding user is very easy: 1. Select Users Module 2. Fill User data in editor shown on Fig 8C 3. Click "Add new".

Fig. 8C: Edit user Only when users is saved, You could assign permissions to him. On list in the left edge of editor, You could check which options User could use. As You see this menu is exactly the same as Tree menu in main window (see 7). If "Add new" button is disabled is problably because of "Assign permissions (only for saved users)"

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is checked. When assigning permissions, You could not add new users - because users has to be saved and has unique identifier (generated during saving) to assign system permissions. New added user problably should doesn't has permissions to use Users module, because in other case, he would to change their self permissions .... Notice: If You add at least one user application will always ask for login data on startup. It's good habit to have one superuser that has permissions to each options! It's possible to add only one user with limited permissions to some features and to... Users module! Then nobody could change add or remove users. This is dangerous situation. When user has limited permissions, after his login he will see features that he couldn't use grayed, as on figure 8d. !1.0VSMDisabledRights.jpg|align=center! Fig. 8D: Some disabled features for limited user.

Tariffs comparer Tariffs comparer module will help You to compare tariff rates with similar or same prefixes from different tariffs. You could use it for example to check wich reseller offer cheaper calls to the same prefixes, or balance tariffs in Your system. To use this module: 1. Open module using tree menu or Main menu in VSM. 2. Set tariffs You want to compare from comboboxes at the top of window as shown on Fig 9a. 3. You could set multiplier. Multiplier sets compare condition between matched tariff rates - for example You could set it yo 1,0 and operator to = (default is -1 - not use) to see only matched tariff rates that have the same rate. You could filter tariff rates greater than some percentage values using this filter field. It could be very usefull. 4. Click "Apply filter" to see matched prefixes.

Fig. 9A: Tariff comparer control panel. You have to only set which tariffs You wan't to compare, and voice rate mulitplier that will set compare condition between tariffs. As result of comparing tariffs, You will see report with mached rates as on Fig. 9B.

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Fig. 9B: Matched tariff rates. Tariff rates are matched by similarity of prefixes. For example if one tariff there is a prefix 04865 but in second only 048 (more global) - they will be compared. This is only an example. Real algorithm used there is of course more complex.

VSM, VSC, VSR - AutoUpdate In new versions of config's there is tool called AutoUpdate. Using this util You could be always up-to-date with latest software upgrades and issue repairs. AutoUpdate is designed to be as simplest as it's possible. Upgrading application with it takes no more than few minutes and few mouse clicks. Note: It's strongly recommended to use AutoUpdate for checking for updates at least once a week. AutoUpdate tool is the same in VSM, VSC and VSR, only difference between this version is that, that in VSC & VSR shortcut to Autoupdate is placed in Windows Start Menu, and in VSM - in VSM main menu under File submenu. Starting Autoupdate: • VSC and VSR: ° select option from windows Start Menu as shown on fig. A. • VSM ° select option from VSM main menu from File submenu as shown on fig. B.

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Fig. B: Starting AutoUpdate for VSM

Fig. B: Starting AutoUpdate for VSM In VSM Application will ask You If You wan't to close current VSM session and check for updates. You have to answer "Yes" in this dialog (shown on Fig. c). VSM will be terminated and AutoUpdate will be executed.

Fig. C: Do You Wan't to clsoe VSM and check For Updates? Of course answer "Yes". After Launching AutoUpdate. You could see in main window which App will be updated and which is current version of it (on the status bar in the bottom of the main window). See Fig. D to check how main window looks like.

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Fig. D: Main window of AutoUpdate utill. As we've written before - use of AutoUpdate is very simple. All You would Like use features are placed on top command bar - see Fig E.

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Fig. E: Commands to use in AutoUpdate Meaning of these commands: • Show new - check's for new updates for Your version of app and show list with updates in the middle of the window, • Show installed - show's updates You've installed in the past, • Return to application - please use this button to close Update-session and return back to application You've updated. AutoUpdate on start, and after selecting "Show new" command will connect to Update Server and download list of updates currently available for Your version of software. If any updates are available You will see check list as shown on Fig F.

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Fig. F: Available updates. Each update has assigned priority (High, Medium or Low), described changes that was made in this update, and specified file size to downlad. It's recommended to install all updates with "High priority". Other updates could wait. Often, newer updates covers some functionalities of older updates, but it's recommended to install all updates instead of only last one. After reading updates description please check which updates You wan't to install using checkboxes on the left (as shown on Fig. F). After selecting, please perform Install checked command under the updates list. Application will download, unpack and install each selected updates. This could take some time, and progress of this operation You could see on progress in the bottom of the window. You could cancel installation of updates every time. But updates which were installed will not be removed.

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Fig. G: Installation progress. After installation of updates You could see message box with text "Installtion Successfull". In other case, there some problems occurs durring installation. You will se information dialog with error messages and list of not-installed updates. Of course You could and probably shall to try install this updates again. Note: Each update is downloaded from Updates Server. If You've some problems with network connections this could generate AutoUpdate problems. Please be sure that Internet connection work's fine when You updateinng VSM, VSC or VSR. After installing Updates You could open updated application choosing "Return to application" option from toolbar. Note: All changed files are backup-ed before installation of updates. If You get some problems after installing updates, You could always restore changed Files. To do this, enter application folder / VSM / Backup (or /vsc/backup, /vsr/backup) - for example this could be: • for VSM: C:\Program Files\VoipSwitch\VoipSwitch 2.0\VSM\Backup • for VSR: C:\Intepub\wwwroot\vsr\backup • for VSC: C:\Intepub\wwwroot\vsc\backup. In those folders You have subfolders names as the date when update was installed for example enter subfolder: "2007-08-13_11721" to get backup of files replaced on 2007-08-13 - 11:07:21. You could copy these files to main application directory to recover state before selected update.

Search client's

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Sometime's You wan't to know which type has client with specified login. Usually, when client gives You a login, but don't tell anything about type. For this situations there is feature "Search.." (see fig. a) button in main tool bar in VSM:

Fig. A: Search clients button and dialog. This command will search for specified login in all client types and open specified client-type's lists with filtered logins to show You all matched records.

Validate prefixes In VSM, VSC in Resellers. Validate prefixes command allows You to check prefixes structure for selected reseller. It checks if sub resellers and client's resellers are consistent with their parent's. Sometime's when client's are imported, and by the time parent's prefix was changed - then could be situation that prefixes aren't consistent. This feature will check and print all clients and subresellers that should be repaired as on figure a.

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Fig. A: Validating prefixes. All clients and resellers with inconsistent prefixes are printed out.

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