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Nuevo concurso para España, Argentina, Brasil, Chile, Colombia, Ecuador, México, Perú y Portugal.

Elastix

Centralita telefónica en red. PBX. VoIP.
- Juan Antonio Villalpando -

Volver al índice del tutorial

____________________________

- Troncal a línea telefónica.

- Troncal a una red telefónica mediante Grandstream HT503

Adaptador Telefónico Analógico (ATA), es un enrutador o puerta de enlace hacia ethernet o línea telefónica PSTN. Lo podemos encontrar por 65 €.

Mediante este dispositivo podemos conectar nuestro Asterisk a una línea telefónica para poder enviar y recibir llamadas desde el exterior. Puede enrutar tanto a líneas de ethernet como a líneas telefónicas.

http://www.voipandgo.be/grandstream-ata503.html

ht_quickstartguide_spanish.pdf

ht503_usermanual_english.pdf

ht70x_50x_quick_user_guide_spanish.pdf

grandstream503_usermanual.pdf

 

Conexiones.

- Salida de llamadas: Queremos que cuando en un Teléfono IP interno pulsemos 9 666 67 78 89, la llamada salga por la línea teléfónica y conecte con ese teléfono externo.

- Entrada de llamadas: Cuando alguien desde el exterior llame a nuestro número 812 34 56 78, la llamada la gestione el Asterisk y automáticamente entre un IVR que tenemos confiturado en Asterisk, para que pueda elegir la Extensión interna de contacto.

NOTA: si necesitaramos más terminales telefónicos, utilizaríamos un Switch. La conexión de Internet, que vendría de un Router, iría al conector WAN. Y luego conectaríamos la conexión LAN con el Switch.

- Creamos el Troncal

- Troncal: ht503

Detalles del par:

host=192.168.1.223
type=peer
canreinvite=no
insecure=very
dtmfmode=rfc2833
quality=yes
port=5062

----------------------------------

Detalles del usuario:

context=from-trunk
host=dynamic
insecure=very
type=friend
dtmfmode=rfc2833
secret=1234aa
nat=yes

----------------------------------

- Ruta saliente: ht503_saliente

Cuando presionemos un número que comience por 9 y tenga 9 cifras (XXXXXXXXX), el curso de la llamada se desviará al troncal ht503, tomará los datos de Opciones salientes y se dirigirá al 192.168.1.223 que es el dispositivo HT503 para que tome la llamada y la saque al exterior por la línea telefónica. Es decir, si queremos llamar al 666 67 78 89, debemos teclear 9 666 67 78 89. El 9 inicial indicará al Asterix que nos queremos comunicar mediante el troncal ht503.

En caso que no tenga éxito la conexión, entrará en Normal Congestión y saldrá una voz diciendo "Todas las líneas están ocupadas".

Normal Congestión: cuando no conecta sale la voz diciendo... todas las líneas están ocupadas...

- Ruta entrante: ht503_entrante

Cuando llegue una llamada del exterior, entrará en el dipositivo HT503, que la enviará al Servidor Elastix 192.168.1.222 y entrará en el usuario ht503_la_entrada (1234aa) que está configurado en el troncal (Opciones entrantes / Contexto del Usuario / Detalles del Usuario).
Si la conexión tiene éxito Establecerá el destino del IVR Festivos.

Cuando entre la llamada ira a un IVR, saldrá una voz diciendo algo así... Bienvenido. Si quiere hablar con el departamento de ventas pulse 1, si quiere hablar con el departamento de producción pulse 2, si quiere hablar con un operador pulse 3.
Consulta en un apartado anterior de este tutorial cómo se construye un IVR.

_________________________________________
_________________________________________
_________________________________________

- Ahora configuramos el GrandStream HT503

Entramos en un navegador web y escribimos la IP del GrandStream HT503, en mi caso 192.168.1.223

En caso que no sepamos la IP conectamos la entrada WAN del GrandStream a nuestro Router y luego consultamos las conexiones del Router.



Las contraseñas siempre las dejamos en: admin

 

_______________________

- Basic Settings

Le asignamos al GrandStream HT503 una IP estática, le establecemos la máscara y puerta de enlace de la red a la que está conectado.

IP: 192.168.1.223
Máscara: 255.255.255.0
Puerta de enlace: 192.168.1.1

- Cuando entre una llamada irá al Servidor Asterisk, usuario 1812345678 (Número DID de la Ruta entrante)
- La IP del Servidor donde tenemos Asterisk: 192.168.1.222

- Advanced Settings

Configuramos los tonos de llamadas de España:
(es IMPORTANTE configurar los tonos, si no lo haces saldrá el mensaje de "Todas las líneas están ocupadas" )

Dial Tone: f1=425@-10,f2=0@-10,c=0/0;
Ringback Tone: f1=425@-10,f2=0@-10,c=150/300;
Busy Tone: f1=425@-10,f2=0@-10,c=20/20;
Reorder Tone: f1=425@-10,f2=0@-10,c=20/20-20/20-20/60;
Confirmation Tone: f1=350@-11,f2=440@-11,c=100/100-100/100-100/100;
Call Waiting Tone: f1=440@-13,c=300/10000-300/10000-0/0;

- Para otros países consultar el final de esta página.

Identificación de tonos de otros países.

- FXS Port

- Esto se utilizar para conectar al HT503 un teléfono analógico para que se pueda integrar en la conguración del Asterisk. En nuestro caso no utilizaremos tal teléfono analógico, asi que Desactivamos esta cuenta.

- FXO Port

Establecemos la IP del Servidor Asterisk, en mi caso 192.168.1.222
y el usuario de entrada troncal: ht503_la_entrada
Authenticate ID:
ht503_la_entrada
Authenticate Password: 1234aa

Ponemos tono de desconexión con la configuración de España:

Enable PSTN Disconnect Tone Detection: Yes
PSTN Disconnect Tone: f1=425@-10,f2=0@-10,c=20/20;

En Dial plan: { xxxxxxxxx }       

En AC Termination model, marca Impedance-based    

En Wait for Dial Tone: Yes

______________________________

Colombia:

Dial Tone: f1=425@-11,f2=425@-11,c=0/0;
Ringback Tone: f1=425@-11,f2=425@-11,c=100/450;
Busy Tone: f1=425@-11,f2=425@-11,c=25/25;
Reorder Tone: f1=425@-11,f2=425@-11,c=25/25;
Confirmation Tone: f1=350@-11,f2=440@-11,c=10/10;

Argentina: http://downloads.3cx.com/downloads/misc/GXPblog/CountryToneSetValues.pdf

Dial Tone: f1=425@-25,f2=425@-25,c=0/0;
Ringback Tone: f1=425@-25,f2=425@-25,c=1000/4500;
Busy Tone: f1=425@-11,f2=425@-11,c=370/320;
Reorder Tone: f1=425@-11,f2=425@-11,c=370/320;
Confirmation Tone: f1=425@-10,f2=425@-10,c=100/100;

Para otros países consulten:

Identificación de tonos de otros países.

y adaptenlo a Grandstream de esta forma:

(Síntaxis: f1=freq@volumen, f2=freq@volumen, c=on1/off1-on2/off2-on3/off3; [...])

(Nota: frecuencia: 0 - 4000Hz; volumen: -30 - 0dBm)

(Default: Busy Tone: f1=480@-24,f2=620@-24,c=500/500;)

México:

Busy tone - 425 0.25 on 0.25 off ===================== f1=425@-10, f2=425@-10, c=25,25;
Congestion tone - 425 0.25 on 0.25 off ================ f1=425@-10, f2=425@-10, c=25,25;
Dial tone - 425 continuous ========================= f1=425@-11,f2=425@-11,c=0/0;
Ringing tone - 425 1.0 on 4.0 off ===================== f1=425@-11,f2=425@-11,c=100/400;
Warning tone - operator intervening 425 0.5 on 0.17 off 0.17 on 0.17 off ==== f1=425@-11,f2=425@-11,c=50/17-17/17;

______________________________

 

Glosario.

* ADSL - Asymmetric Digital Subscriber Line . Modems attached to twisted pair copper wiring that transmit from 1.5Mbps to 9Mbps downstream (to the subscriber) and from 16kbps to 800kbps upstream, depending on line distance.

* ARP - Address Resolution Protocol is a protocol used by the IP ( Internet Protocol ), IPv4, to map IP network addresses to the hardware addresses used by a data link protocol. The protocol operates below the network layer as a part of the interface between the OSI network and OSI link layer. It is used when IPv4 is used over Ethernet.

* ATA - Analogue Telephone Adapter . Enables analogue telephone to be used in data network for VoIP.

* CODEC - Abbreviation for Coder-Decoder . It is an analog-to-digital (A/D) and digital-to-analog (D/A) converter for translating the signals from the outside world to digital, and back again.

* DATAGRAM - A data packet carrying its own address informaton so it can be independently routed from its source to the destination computer.

* DNS - Short for Domain Name Service , an Internet service that translates domain names into IP addresses.

* DSP - Digital Signal Processor . A specialized CPU used for digital signal processing. All Grandstream products have DSP chips inside.

* DTMF - Dual Tone Multi Frequency . The standard tone-pairs used on telephone terminals for dialing using in-band signalling. The standards define 16 tone-pairs (0-9, #, * and A-F) although most terminals support only 12 of them (0-9, * and #).

* FXO - Foreign eXchange Office . An FXO device can be an analog phone, answering machine, fax, or anything that handles a call from the telephone company.
An FXS interface will accept calls from FXS or PSTN interfaces. All countries and regions have their own standards.
FXS is complimentary to FXS (and the PSTN).

* FXS - Foreign eXchange Station . An FXS uses additional hardware to generate the ring signal to the FXS extension (usually an analog phone).
An FXS device will allow any FXS device to operate as if it were connected to the phone company. This makes your OBX the POTS+PSTN for the phone.
The FXS interface connects to FXS devices (by an FXS interface, of course).

* DHCP - The Dynamic Host Configuration Protocol is an Internet protocol for automating the configuration of computers that use TCP/IP. DHCP can be used to automatically assign IP addresses, deliver TCP/IP stack configuration parameters such as the subnet mask and default router, and to provide other configuration information such as the addresses for printer, time and news servers.

* ECHO CANCELLATION - Echo Cancellation is used in telephony to describe the process of removing echo from a voice communication in order to improve voice quality of a telephone call. In addition to improving quality, this process improves bandwidth savings achieved through silence suppression by preventing echo from travelling across a network. There are two types of echo of relevance in telephony - acoustic echo and hybrid echo. Speech compression techniques and digital processing delay often contribute to echo generation in telephone networks.

* H.323 - A suite of standards for multimedia conferences on traditional packet-switched networks.

* HTTP - Hyper Text Transfer Protocol . The World Wide Web protocol that performs the request and retrieve functions of a server.

* IP - Internet Protocol . A packet-based protocol for delivering data across networks.

* IP-PBX - IP-based Private Branch Exchange .

* IP Telephony ( Internet Protocol Telephony , also known as Voice over IP telephony ). A general term for the technologies that use the Internet Protocol's packet-switched connections to exchange voice, fax, and other forms of information that have traditionally been carried over the dedicated circuit-switched connections of the Public Switched Telephone Network (PSTN). The basic steps involved in originating an IP Telephony call are conversion of the analog voice signal to digital format and compression/translation of the signal into Internet Protocol (IP) packets for transmission over the Internet or other packet-switched networks; the process is reversed at the receiving end. The terms IP Telephony and Internet Telephony are often used to mean the same; however, they are not 100 per cent interchangeable, since Internet is only a subcase of packet-switched networks. For users who have free or fixed-price Internet access, IP Telephony software essentially provides free telephone calls anywhere in the world. However, the challenge of IP Telephony is maintaining the quality of service expected by subscribers. Session border controllers resolve this issue by providing quality assurance comparable to legacy telephone systems.

* IVR - IVR is a software application that accepts a combination of voice telephone input and touch-tone keypad selection and provides appropriate responses in the form of voice, fax, callback, e-mail and perhaps other media.

* NAT - Network Address Translation .

* PPPoE - Point-to-Point Protocol over Ethernet is a network protocol for encapsulating PPP frames in Ethernet frames. It is used mainly with cable modem and DSL services.

* PSTN - Public Switched Telephone Network . The phone service we use for every ordinary phone call, or called POTS ( Plain Old Telephone Service ), or circuit switched network.

* RTCP - Real-time Transport Control Protocol . With the RTP it is delivery and packaging of multimedia data, but does not transport any data itself. It is used periodically to transmit control packets to participants in a streaming multimedia session. The primary function of RTCP is to provide feedback on the quality of service being provided by RTP.

* RTP - Real-time Transport Protocol defines a standardized packet format for delivering audio and video over the Internet.

* SDP - Session Description Protocol is a format for describing streaming media initialization parameters.

* SIP - Session Initiation Protocol . SIP is a text-based protocol suitable for integrated voice-data applications. SIP is designed for voice transmission and uses fewer resources while it is considerably less complex than H.323. The Grandstream products are SIP-based.

* STUN - Simple Traversal of UDP over NAT is a network protocol allowing clients behind NAT to find out its public address, the type of NAT it is behind and the internet side port associated by the NAT with a particular local port. This information is used to set up UDP communication between two hosts that are both behind NAT routers. It works with non-symmetric NAT routers.

* TCP - Transmission Control Protocol is one of the core protocols of the Internet protocol suite. Using TCP, applications on networked hosts can create connections to one another, over which they can exchange data or packets. The protocol guarantees reliable and in-order delivery of sender to receiver data.

* TFTP - Trivial File Transfer Protocol , is a very simple file transfer protocol, with the functionality of a very basic form of FTP. It uses UDP (port 69) as its transport protocol.

* UDP - User Datagram Protocol is one of the core protocols of the Internet protocol suite. Using UDP, programs on networked computers can send short messages known as datagrams to one another. UDP does not provide the reliability and ordering guarantees that TCP does; datagrams may arrive out of order or go missing without notice. UDP is faster and more efficient for many lightweight purposes.

* VLAN - A Virtual LAN , is a logically-independent network. Several VLANs can co-exist on a single physical switch.

* VoIP - Voice over the Internet Protocol . VoIP encomprasses many protocols. All the protocols do some form of signalling of call capabilities and transport of voice data from one point to another.

http://www.3cx.es/topic/tonos-marcado-e-identificacion-de-llamadas-ht503-espana/

_________________________________________________

 

- Mi correo:
juana1991@yahoo.com
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