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2.

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

Limit ring time Maximum ring time in seconds: [30]

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.

Use media timeout Choose media timeout value in seconds: [20]

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.

Refuse connection when destination rate is higher than client rate

Limit call duration Maximum call duration in minutes: [59]

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

Limit number of hops (re-routing policy) Maximum number of hops:

[-1]

Guest account:

-- not used --

Authorization

Default value

Authorize incoming calls

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.

Users can log by IP number

Authorize by ANI (Calling Party Number)

opcin global (usar con cuidado) que permite a todos los clientes que se autoriz por primera vez por ANI / CLI.

Use custom password policy for web access Default value

permite a los clientes tener una contrasea diferente para el dispositivo y el acceso a la web del portal. Description

Use retail clients

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

Use new codecs

not used, must be ticketed by default.

Use time spans in DP

enables the time spans feature in dialing plan.

Use load balancing

enables the load balancing feature in dialing plan.

Use tariff plans

enables the tariff plans feature. defines the starting local UDP port used to send voice media (RTP) packets from.

RTP start port:

[6000]

2.1.1.2 Failed calls


Por defecto VoipSwitch no guarda las llamadas fallidas para los eventos siguientes. Se ha diseado de esta manera para evitar el recuento de dichas llamadas para las estadsticas generales, como por ejemplo ASR. Esta opcin tambin ayuda a evitar que la tabla de base de datos callsfailed crezcan demasiado rpidamente. Por favor, seleccione algunas opciones slo cuando se requiere en su configuracin de servicio. There was no destination in dialing plan Client had no money Destination was offline Limit was reached Dialed number was disabled There was no tariff Source and destination have no same codecs

There was no destination in dialing plan

Client had no money

Destination was offline

Limit was reached

Dialed number was disabled

There was no tariff

Source and destination have no same codecs

2.1.1.3 Rerouting and ending calls

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

2.1.1.5 Cluster configuration


Esta opcin slo se utiliza cuando voipswitches varios usuarios trabajan como un grupo conectado a una base de datos comn. En tal caso, cada VoipSwitch debe tener un conjunto nico de identificacin.

2.1.2 VoipSwitch listeners


The whole procedure is described in detail in the video tutorial VSM - Settings and Listeners Configuration Primero abrir el VoipSwitch Manager haciendo clic en el botn Inicio y, a continuacin, abra Todos los programas, y abra la carpeta VoipSwitch. A continuacin, seleccione Voipswitch Manager. A continuacin expanda el elemento de men Configuracin y seleccione VoipSwitch. Se abrir el mdulo de configuracin VoipSwitch Hay 5 pestaas: Echemos un vistazo a cada uno de ellos. Tenemos que configurar una serie de campos de esta ficha. Compruebe el anillo Lmite casilla de hora Y establecer

el lmite de anillo tiempo mximo en segundos a 30 segundos


Este valor debe estar configurado para configurar el evento Respuesta NO en respuesta reglas La opcin Usar los medios de comunicacin Tiempo de espera Establece el tiempo en segundos para desconectar la llamada si una de las partes que llaman deja de enviar paquetes de medios. El valor predeterminado es de 20 segundos. Si el "Rechazar conexin cuando la tasa de destino es mayor que una velocidad de cliente" se comprueba que impide la conexin de llamadas con una ganancia negativa. El llamado lmite de duracin casilla necesita ser comprobada. Esto define la duracin de la llamada en minutos mximo despus de lo cual cada llamada se desconectar por VoipSwitch. Previene algunos casos aleatorios cuando la llamada no ha sido terminada por ambos lados. Puede establecer este a 60 minutos. El lmite en el nmero de casilla lpulo necesita ser comprobada. Esto limita el nmero de intentos de reencaminamiento para hacer coincidir los prefijos en el plan de marcacin. Establecer el nmero de saltos a 5. La siguiente pestaa se falla llamadas. Deje en blanco las entradas. VoipSwitch no guarda las llamadas fallidas por defecto para los eventos siguientes. La siguiente pestaa se desvo y finalizar llamadas. Deje esta intacta. Vaya a la pestaa de grabacin llamadas activas. Compruebe las llamadas activas guardar en la caja de DB. Esto es necesario para la opcin de llamadas activo en VSR, VSC, llamadas Monitor y los mdulos de locutorio VSPortal. Asegrese de que las llamadas activas conectadas estado est activada. Esta opcin y los otros slo son utilizados por la aplicacin Monitor de llamadas. La siguiente pestaa es para configuracin de clster. Deje esta intacto y haga clic en Guardar cambios. A continuacin, haga clic en Aceptar. Para configurar los oyentes VSM, expanda el elemento de men y vaya al SIP. El mdulo tiene dos fichas: Configuracin SIP principales y oyentes. En la ficha Principal SIP Settings VoipSwitch entrar en el campo reino. A continuacin, vaya a la pestaa de Listeners. Aqu es donde se puede editar y aadir las direcciones IP. Desde la ventana de SIP disponibles oyentes, seleccione la direccin IP de la lista y haga clic en el botn de flecha Derecha. La direccin IP se desplaza a la ventana seleccionada direcciones. Puede modificar el protocolo tcp o udp. En general, el protocolo UDP est seleccionado. Puede agregar y IP adicional en la lista disponible al resaltar la direccin IP de la lista y hacer clic en la flecha derecha. Se puede seleccionar un protocolo diferente para una IP. Resalte la IP que desea editar y seleccione el protocolo - en este caso tcp.

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

2.2 Clients configuration


En general el procesamiento del flujo de llamada VoipSwitch de un cliente a un destino a travs del Plan de Marcacin. Cada llamada que viene a VoipSwitch est autorizada antes de la transformacin y se asigna a una cuenta de cliente determinado. Cuentas de los clientes son diferentes en sus funciones, el mtodo de opciones de autorizacin y disposicin. Algunas de las caractersticas son comunes para todo tipo de clientes.
Client types and authorization methods Clients Authorization Description utiliza sobre todo para los transportistas en los servicios mayoristas, cabinas de locutorio y para autorizar nmeros DID se utiliza para: activar una devolucin de llamada llamar a los escenarios de IVR llamar a los clientes minoristas y cobrarles para las llamadas entrantes (anteriormente llamado gateway / gw clientes)

Wholesale clients

IP Auth. Prefix H323 ID

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

2.3 Destination endpoints configuration


En General el proceso de Flujo de Llamada es de un cliente a un destino a travs del dialing Plan (plan maestro de numeracin)

Existen los siguientes tipos de destino disponibles para la configuracin:


Destination Gateways Description a termination, carrier side, supplier where calls are sent based on provided ip or dns address a termination, carrier side, supplier where calls are sent based on provided ip or dns address however this end point must be registered with login/password. used for routing calls to the clients registered to the same voipswitch server or cluster. DNS server which changes the number to SIP/H323 URI (address) of the number's owner. The most known enum route are e164.arpa and e164.org a customized database query triggers callback calls IVR module for providing voice messages and predefined call scenarios.

GK/Registrar

Retail clients

Enum routes Lookups Callback routes VoipBox

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 Working with the dialing plan


Working with the Dialing Plan is fully described in the VSM Manual, in the chapter 1.2.4 Dialing plan. The subsections below expand on the description from the VSM manual.

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

2.4.1 Call flow


The general call flow processing is from a Client to a Destination through the Dialing Plan. 1. Una llamada entrante est autorizado en contra de toda configuracin de los clientes (Fig. 1) 2. The maximum call duration is calculated based on the destination number and the authorized client tariff. (Fig. 2) 3. Routing - looking for the best match in the Dialing plan. (Fig. 3) 4. The call is sent to the destination endpoint. (Fig. 4)

2.4.2 Route types


Route types are the same as the destination endpoints described in the section: 2.3 Destination endpoints configuration Gateways a termination, carrier side, supplier where calls are sent based on provided ip or dns address GK/Registrar a termination, carrier side, supplier where calls are sent based on provided ip or dns address; however, this end point must be registered with a login/password. Retail clients used for routing calls to the clients registered to the same voipswitch server or cluster. Enum routes DNS server which changes the number to SIP/H323 URI (address) of the number's owner. The most known enum routes are e164.arpa and e164.org Lookups customized database query Callback routes triggers callback calls VoipBox IVR module for providing voice messages and predefined call scenarios. Groups (of Retail clients)

Hunt groups of Retail clients

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..

2.4.4 Load sharing (balancing)


Load sharing is a functionality enabling controlled traffic distribution among different endpoints (gateways/gatekeepers or carriers). A balance share value can be assigned to each route, which represents the percentage of the total traffic sent to a given destination. Voipswitch will control the incoming traffic and use a sophisticated algorithm to achieve the distribution closest to the projected. For example if we set even distribution 33% per each of three gateways handling calls to, for example, the UK (44 country code), the voipswitch will direct the incoming calls to defined routes so that when at any moment we run a report we should see that the exact (or very close to it) number of calls has been sent through each route. Note that all priorities are the same for each route included in the load balancing; this is because in this scenario each route shares the same level of importance (same quality or same price etc.). There are no limits to the number of sections utilizing the load sharing functionality. For example, there can be different groups for UK traffic, for the US and so on.
Configuration procedure:

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%.

2.4.5 Least call routing (LCR)


One of the purposes of the traffic failover functionality, described earlier, is to maximize profits by sending calls first to carriers with the lowest price (which often goes in pair with lower quality) and then in case of failure tries other routes with worse pricing. This however requires us to change the priorities depending on the changes in carrier's rates sheets. The process quickly becomes complicated as we have to compare rates for the interesting us routes, pick the cheapest and change the dialing plan. The Least Call Routing (LCR) mechanism is intended to help us. Its task is to check the cost tariffs assigned to the carriers that belong to LCR group and compare their rates. This is performed every defined interval. When the rate has changed it is reflected in the

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 Main scenarios configuration


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

2.5.2 Retail SIP services


Usando Voipswitch Platform, un proveedor de servicios puede fcilmente desarrollar toda una gama de servicios minoristas para residencial y clientes de negocios (Fig. 1, 2). Clientes Minoristas pueden hacer y recibir llamadas con SIP device o softphone (including mobile softphones). La misma cuenta se puede utilizar con tarjetas de llamadas y servicios callbacks asi como la contrasea es tambin el nmero PIN. Los nmeros de telfono virtuales es un servicio que permite a los clientes a alcanzar directamente en sus propios nmeros de telfono de los abonados de otras redes, incluyendo PSTN - lneas regulares de telecomunicaciones.VoipSwitch has integrated major DID providers APIs allowing to offer virtual numbers to end users using Customer Portal.
Making/receiving calls from SIP adapaters Virtual numbers DIDs

configuring DID providers, configuring web, defining pricelist local DIDs *free DIDs e.g. free E164 numbers

Llamadas gratis entre usuarios


using DIDs (dids numbers rates set to 0 in users tariff) using Logins

Calling using ENUM Allowing anonymous inbound calls

(e.g. calls from other providers, E164 internet numbering plan)


Auto-provisioning - remote configuration

2.5.3 Calling cards

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

Calling Cards service number (DID) 442081368999

End-user1 Caller ID (ANI) End-user1 (IVR client) login End-user1 PIN (password)

1234567890 testpin 3210

End-user2 Caller ID (ANI) End-user2 (Retail client) login End-user2 PIN (password)

9876543210 9876543210 5521

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 prefixes for the service number (Tariff)

Verify if the Tariff (Fig. 4) assigned to the wholesale client "Local_DID" includes the service number or prefix.

Add the Calling Cards service number (Dialing Plan)

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.

Add end-users (IVR or Retail clients)

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.

Pinless - ANI (Caller ID) authorization

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).

Registering ANI (CallerID):

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

See 6.2.7 Speed dial


Recharge

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

Charging for calls depending on Caller ID

(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)

See the link: Select language + ... scenarios

Account expiration and lot generation

See the link: 1.2.1.10 Account Generation


One stage scenario

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