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Windows registry settings are overwritten by the voipswitch_config.xml file located in the
folder:
c:\Program Files (x86)\VoipSwitch\VoipSwitch 2.0
.Database connection
VoipSwitch settings
VoipSwitch listeners
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Main settings
Call settings
Default
[30]
[20]
Refuse connection
when destination rate
is higher than client
rate
Description
[59]
[-1]
Maximum number of
hops:
Guest account:
-- not
used --
Authorization
Default
value
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permite la autorizacin de todas las llamadas que llegan al
servidor.
Authorize incoming
calls
Authorize by ANI
(Calling Party
Number)
Description
Use resellers
defines the starting local UDP port used to send voice media
(RTP) packets from.
[6000]
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Client had no money
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Dialed number was disabled
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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.
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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.
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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
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
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services
PBX clients Login&password
PBX subaccounts
CallShop
clients
Login&password
CallBack
clients
Login&password
used only for ANI callback services
Caller ID (ANI)
IVR clients
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
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Description
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
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 route are e164.arpa and e164.org
Lookups
Callback routes
VoipBox
IVR module for providing voice messages and predefined call scenarios.
Each Destination type is described in details at the VSM manual: 1.2.3 Destinations
chapter
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The subsections below expand on the description from the VSM manual.
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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.
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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%.
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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.
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This page describes how to setup a simple Calling cards scenario based on the following
example:
1234567890
testpin
3210
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End-user2 Caller ID (ANI)
9876543210
9876543210
5521
Configuration steps
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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.
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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.
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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
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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 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