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1 Configuracion Inicial
La configuracin inicial VoipSwitch servidor se debe realizar utilizando la aplicacin VoipSwitch Manager (VSM) Tiendas VoipSwitch ajustes en varios lugares, incluyendo las siguientes: base de datos, del Registro de Windows, aplicacin xml, por ejemplo, voipswitch_config.xml.
Windows registry settings are overwritten by the voipswitch_config.xml file located in the folder: c:\Program Files (x86)\VoipSwitch\VoipSwitch 2.0
El proceso de configuracin inicial requiere la creacin de los siguientes: . Database connection VoipSwitch settings VoipSwitch listeners
Main settings
Call settings Default Description
Tiempo mximo de timbrado en segundos despus de que una llamada puede ser tanto desviado hacia otro destino (de acuerdo con el plan de marcacin) o Reglas de contestador se activar.
permite interrumpir una llamada, si una de las partes que llaman deja de enviar paquetes de medios. Esta opcin no se utiliza para las versiones VoipSwitch superiores a 985,130. impide conectar las llamadas si el prefijo de destino tarifa de voz es ms alta que la tasa del cliente. Esta opcin puede prevenir en contra de calcular un ingreso negativo para las llamadas.
define la duracin de la llamada mxima expresada en minutos despus del cual la llamada ser interrumpida y se dej caer por VoipSwitch. Puede prevenir algunas situaciones al azar cuando la llamada no ha sido terminada por ambos lados. permite cambiar el plan de marcacin poltica reencaminamiento mediante la limitacin del nmero de saltos efectuados si la llamada no se pueden conectar a destinos consecutivos. Por defecto es ilimitado y todos los prefijos coincidan definidas en el plan de marcacin se utilizan. permite que las llamadas desde los dispositivos no autorizados. La cuenta utilizada para autorizar las llamadas por defecto cuando la llamada entrante no est autorizada ninguna cuenta de otro cliente. Sin esta opcin en todas las llamadas no autorizadas son rechazadas. Description
[-1]
Guest account:
-- not used --
Authorization
Default value
permite la autorizacin de todas las llamadas que llegan al servidor. (para uso futuro - VoipSwitch como un proxy o un convertidor de protocolo) permite a los clientes mayoristas a ser reconocidos por direccin IP.
opcin global (usar con cuidado) que permite a todos los clientes que se autoriz por primera vez por ANI / CLI.
permite a los clientes tener una contrasea diferente para el dispositivo y el acceso a la web del portal. Description
permite a los clientes al por menor autorizacin llamadas. activa la funcin de los revendedores para calcular los costos de revendedores para las llamadas de sus clientes.
Use resellers
enables the tariff plans feature. defines the starting local UDP port used to send voice media (RTP) packets from.
[6000]
2.1.1.4 Active calls recording Save active calls in DB Esta opcin permite VoipSwitch para guardar la informacin de llamadas activas a la tabla de base de datos adicional: currentcalls, que es utilizado por otros mdulos.
Required for the active calls option in VSR, VSR, PBX-Calls monitor and VSPortal Callshop module.
Las siguientes opciones slo son utilizados por la aplicacin de monitorizacin de las llamadas:
Active calls - ringing state Active calls - dialing state Active calls - connected state
Haga lo mismo para Registrador de abajo. Seleccione la IP de la lista disponible y pulse la flecha derecha. Seleccione el protocolo. De nuevo, si usted desea agregar una direccin IP adicional, seleccione la IP de la lista disponible y haga clic en la flecha derecha. Y de nuevo, seleccione el protocolo. Cuando haya terminado su seleccin, haga clic en Guardar cambios y haga clic en Aceptar. Luego vaya a la pgina de H323. La pgina se divide en dos mitades: H323 oyentes, y Gatekeeper. Seleccione la IP H323 oyentes y haga clic en la flecha izquierda. Los movimientos de la direccin IP al ordenador disponible ventana direcciones. Haga lo mismo con Gatekeeper. Resalte la direccin y haga clic en la flecha izquierda. La direccin IP se muestra en la ventana de direcciones de ordenador disponible. Haga clic en Guardar cambios y en Aceptar. Las direcciones IP se puede ver en el mdulo de devolucin de llamada, y el mdulo de Callshop. Cierre el Administrador de VoipSwitch. Abra la aplicacin VoipSwitch haciendo clic en el botn Inicio y, a todos los programas. Abra la carpeta VoipSwitch, y empezar a VoipSwitch. En la ventana de registros de abajo podemos ver la conexin a la base de datos ha sido establecida. Tambin podemos ver que el oyente SIP ha comenzado. Cierre la aplicacin VoipSwitch
Wholesale clients
Retail clients
Login&password (previously called common clients) mainly used for ip phone PIN (password) devices, ATA's and software dialers in retail services, could be Caller ID (ANI) used for PINs and recharge PINs in calling cards and callback
services PBX clients Login&password PBX subaccounts CallShop clients CallBack clients IVR clients used as PBX admin accounts to maintain multi-tenant PBX configurations
Login&password PBX end-users belonging to the same company used for callshop services as main callshop owner accounts which control their cabins
Login&password
Login&password used only for ANI callback services Caller ID (ANI) PIN (password) used as PINs in calling cards and callback services Caller ID (ANI)
Caractersticas Comunes Login/password Calls limit Codecs Prefixes Currency (dinero, moneda) Tariff (tarifa) Remaining funds (balance) Personal data
See more in the VSM manual chapter: 1.2.1.1 Features common for most client types
Adding a client
Los clientes pueden ser agregados y configurados usando el VoipSwitch Manager (VSM), web VoipSwitch Config (VSC) or by a reseller through the VSR web module. In addition an automatic sign-up (client registration) is available through the Portal and OnlineShop modules. Below there are links to video tutorials which show how to add clients in VSM:
VSM - Adding new clients VSM - Adding wholesale clients PBX - New PBX account using VSM
GK/Registrar
Retail clients
Groups (of Retail Hunt groups of Retail clients clients) Destinations should be configured before the Dialing Plan setup so they will be available for selection in the Dialing Plan.
Each Destination type is described in details at the VSM manual: 1.2.3 Destinations chapter
2.4.1 Call flow 2.4.2 Route types 2.4.3 Failover 2.4.4 Load sharing (balancing) 2.4.5 Least call routing (LCR) 2.4.6 Calling among users 2.4.7 Routing based on CLI
2.4.3 Failover
Maximization of the calls completion ratio (ASR) is one of the most important factors in every VoIP deployment. To accomplish this one should secure supplies of the voice termination by arranging contracts with multiple carriers instead of relying on one source only. Voipswitch can work with multiple carriers (gteways/gatekeepers) for each destination, actually there is no limits in number of routes defined for particular code. To specify which route should be taken as first we should use priorities. This parameter can be found in the dialing plan. When adding the first entry for given code, for example 44 like in the picture below, the priority will be set by default to 0. When adding another entry in the dialing plan with the same code (i.e. 44), the priority will change to 1. When adding next it will change to 2 and so on. At any moment we can manually change the priorities and thus the routes order. Just we have to take care of that the priorities differ with each other. If we set the same priorities voipswitch will pick only one route (the first in the database) and will ignore the second with the same priority, unless we enable "balance share" option which is described in the scenario below. The failover procedure starts with voipswitch trying to send a call to the route with priority 0 (or any other number which is highest for given "phone number"). When the remote endpoint has responded with error code or has not responded at all, voipswitch immediately starts trying next route. The whole process lasts untill the call is connected or the last entry with lowest priority failed, only then the release code is sent to the client. For the client there is no indication which route the call has been sent through. In addition we can define on which release codes sent from destinations voipswitch should continue failover procedure and on which not. We can select any SIP or H323 end codes and exclude them from failover (see:.....). Note: Voipswitch automatically performs failover procedures when there is higher (less detailed) code (phone number) defined in the dialing plan. For example if we have an entry for 44 ("phone number" field) and another entry for more general code 4, voipswitch will be re-trying always when the route for 44 fails. To avoid this use "special properties" selector (in the dialing plan) and choose "do not jump" option. It will cause that voipswitch stops on this route.
Configuration procedure:
The scenario can be implemented for any type of client and any type of destination (both gateways and gatekeepers/sip registrars are supported). About adding clients and termination endpoints in wholesale scenarios see above scenarios and related configuration procedures. Go to the dialing plan, add routes that will share the traffic, specify the "phone number", the same for each route, for example country or area code, like 44 in the example above.
For each of the newly created entries in the dialing plan (routes) set different priority, for example start with 0 which is the highest priority and then for subsequent routes 1, 2..
1. The scenario can be implemented for any type of client and any type of destination (both gateways and gatekeepers/sip registrars are supported). See above scenarios and related configuration procedures for information on adding clients and termination endpoints in wholesale scenarios. 2. Go to the dialing plan, add routes that will share the traffic, specify the "phone number", the same for each route, for example country or area code, like 44 in the example above. 3. For each of the newly created entries in the dialing plan (routes) set the same priority, for example 0, which is the highest priority but it can also be any of the lower priorities if you want the group to take the overflow traffic (failing from the routes with the higher priority). 4. Set the "balance share" parameter for each route. This is the percentage of the total traffic. The total percentage, if you sum the values, should equal 100%.
dialing plan - the LCR service changes priorities and thus changes the failover sequence. What is left to us is only to upload new rates sheet when it comes from carrier.
2.5.2 Retail SIP services 2.5.3 Calling cards 2.5.4 Callback services 2.5.5 Fax server setup 2.5.6 SMS configuration
configuring DID providers, configuring web, defining pricelist local DIDs *free DIDs e.g. free E164 numbers
using DIDs (dids numbers rates set to 0 in users tariff) using Logins
Tarjeta de llamadas (Fig. 1, 2) is a popular VoIP service offering the possibility of making calls from regular phones or mobiles through access numbers. DID numbers are offered by DID providers and local telecommunication companies. Every call coming from a DID provider is processed by VoipSwitch with the IVR module.
A user can select one of the available access numbers to use. Each access number can point to a scenario in a different language; for example, a number from Madrid can be operated by the IVR in Spanish
This page describes how to setup a simple Calling cards scenario based on the following example:
The Calling Cards service is available under the number: 442081368999. Calls come from the DID provider called Local_DID. The end-user who will use the calling cards service has Caller ID: 1234567890. Example parameters
End-user1 Caller ID (ANI) End-user1 (IVR client) login End-user1 PIN (password)
End-user2 Caller ID (ANI) End-user2 (Retail client) login End-user2 PIN (password)
Configuration steps
Add a local DID provider (Cliente al por Mayor) Add the Wholesale client with login "Local_DID" (Fig. 3) in order to enable calls from the DID provider or carrier. An IP address (eg. 77.253.221.24) must be added to authorize the provider's incoming connections. Marcar the PIN source option que permite al wholesale client llamar PIN scenarios available in the VoipBox module. This option is required for the calling cards service.
Verify if the Tariff (Fig. 4) assigned to the wholesale client "Local_DID" includes the service number or prefix.
Add the Dialing Plan number: 442081368999, select the Route type: "VoipBox", the available Route below and set up the PIN scenario (Fig. 5). It is possible to choose another IVR scenario which has its name starting with "PIN ...". Each of them is described in the 4.2.1 Built-in scenarios section.
End-users which can use the Calling Cards service are defined as IVR or Retail clients. Add the IVR client with login "testpin" and password: 3210 (Fig. 6). The password is the client's PIN so it must be defined only with digits which allows entering the PIN from a telephone keypad when asked by the IVR PIN scenario.
To avoid entering the PIN the client can have ANI numbers (Caller IDs) defined (Fig. 6). In this case he is only asked to enter the destination number he would like to be connected with when he calls from the number 1234567890. The other way to automatically authorize the end-user is to add the client (9876543210) with the same login as his ANI (Caller ID) and tick the option "Recognize when login=ANI" (Fig. 7).
after entering PIN for the first time - after successful PIN authorization the ANI is registered. It requires setting up one of the PIN scenarios including the word register (Fig. 8) in the Dialing Plan, eg. "PIN + account + time + register" or "PIN + register + only once". See more details in 4.2.1 Built-in scenarios. through Portal - see the link: 6.2.12 Authorized caller IDs by sending SMS - see the link: 2.5.4.2 SMS Callback service
Speed dial
A calling Card service end-user can recharge (top up) their account using a voucher (with a recharge pin) or by using another account balance and its pin. Recharge pins are described in 1.2.1.15 Recharge pin packs whilst using another account balance requires the option Allow to use client's accounts to recharge to be enabled. Both methods: Recharge pins and other accounts pin can be provided to the system
via IVR - requires setting up the "PIN + Recharge" scenario in the Dialing Plan via web - see the link: 6.2.14 Recharge via SMS - see the link: 2.5.4.2 SMS Callback service
(eg. different rate when called from mobile) See the link: 1.2.6.5 Tariffs to ANI
Charging for calls depending on DNIS
(eg. different rate when called through local access number, different when through toll free 0800) See the link: 1.2.6.4 Tariffs to DNIS
Working with languages (selection, language per access number)
The Calling cards service can be also configured to work with the one stage scenario (Fig. 9). This scenario is often used to authorize incoming calls based on a registered ANI. If the ANI is not registered then the caller is asked to enter a PIN. In this case the end user is not asked for a destination number which is taken from the incoming number. See configuration details in the link: 4.2.1 Built-in scenarios