VoipSwitch manual
This page last changed on Feb 25, 2008 byadmin.
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.
Point 1.0 of manual describes VoipSwitch.exe itself and next points about VSM.
Calls in Voipswitch are coming fromClientstowards Gateways or GK/Registrars. SoClientssend calls to Voipswitch while GatewaysandGK/Registrarsterminate 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 asClientsfrom one of 6 available types.Calls, statistics and other reportsare availalble in few parts of VSM.Invoicesare self descriptive butServicesare 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 inGateways. Account inGK/Registrarto register VoipSwitch to gatekeeper or SIP Proxy.
2. Rules how to send calls depending on dialed numbers are defined inDialingPlan.
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 namedTariffs
4. Createclientsto 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 inServices
1.0 Main system
This page last changed on Feb 23, 2008 byadmin.
• 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.
Fig.1 VoipSwitch calls processing window
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. 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
• (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.
• Reload settings mean, that VoipSwitch will read and apply allchanged 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:
Fig.6 VoipSwitch Statistics window.
• 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
° SIP calls - all calls that was usingSIP protocol
° ASR - answer seizure ratio for incoming/outgoing calls
° ACD -(Average incoming/outgoing call duration)
• Registered users - All currently registered users (more details inRegistered 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.
Fig.7 VoipSwitch registered clients window.
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.
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).
After some changes ingatekeepers 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.
Fig.10 VoipSwitch Gatekeepers window with settings.
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)
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).
Fig.11 VoipSwitch Users window.
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) Client offline (disconnected)
2.0 Clients
This page last changed on Feb 24, 2008 byadmin.
• 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
2. PC2Phone clients 3. GK clients 4. Callback Clients 5. IVR clients 6. Common clients 7. Callshop clients 8. 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 describedhere
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 describedhere
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.
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 describedhere.
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
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 inServices.
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.
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
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.
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 describedhere.
Setting termination on PC2Phone client is described indialing plansection 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
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).
Parameterssupported codecs,calls limithave 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 arehot billed
but only when one port of device uses one login and password. Otherwisehot billingthe 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 withVoipTunnel 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 inmanual 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.
• Separatedialing plan ,tariff or caller id prefixcan 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 withIVR 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 torecharge scenarioto add funds using specialrecharge codesor check account state on his account.
Calling from the authorized phones without entering pin. Authorized phones can be added: • on the web,
• 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.
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 inCallshop 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.
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.
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 clientsto 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.
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.
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.
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.
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.
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.
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. Client login
2. Password
3. Tariff ID From system 4. Account state
5. Reseller ID from system 6. 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.
Prefixes
This page last changed on Feb 04, 2008 byrashid.
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.
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.
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
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.
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 optionDon'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,
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
example 6 (adding suffix) Adding suffix (00)
s1*->s1{*}00
exemplary number dialed: 48600789456 will be send as: 4860078945600
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' )
3.0 Destinations
This page last changed on Feb 24, 2008 byadmin.
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 thedialing planrules 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
• 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.
• 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 justone entry in dialing plan. Every such client should have the same beginning.
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 inVoipbox 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 modulesection.
4.0 Dialing plan
This page last changed on Feb 24, 2008 byadmin.
• 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 todestinations. 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.
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.
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!
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 disablefollow mefor 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.
When this option is usedFollow 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.
4.8 Importing and exporting data
4.8.1 Export dialing plan
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
Tech prefix stores value for part defined in VSM asRules 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 definedhere
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.
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.
5.0 Tariffs
This page last changed on Feb 21, 2008 byadmin.
• 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.
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.
