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jueves, 26 de enero de 2012

Configuracion del Granstream HT503 como troncal SIP

 

Aquí les dejo como configurar un Granstream HT503 como troncal SIP con Elastix. En Asterisk  la configuración es prácticamente el mismo procedimiento. Yo particularmente probé la configuración en Elastix y   Asterisk  y me funciono perfecto, pero luego de unos días el aparato empezó a dar problemas  de solo recibir llamadas pero no sacar las llamadas  y recibia el error en Asterisk "Got SIP response 503 "Service Unavailable" luego de actualizar el  firmware y  resetearlo varias veces a la configuración de fabrica funcióno.  De todos modos  si se animan a probarlo muy bien, el aparato mientras esta funcionando es una maravilla pero les recomiendo tener un plan B en caso de que  les presente algun problema como fue  mi caso.









Tutorial on how to make a peer between the HT503 and an Asterisk PBX Server (example of Elastix)

http://mohait86.freehostia.com/gs/?p=31

Posted on March 22, 2011 by admin
Introduction:
This tutorial shows how to configure the HT503 with an Asterisk server without SIP registration. This tutorial is made so you can get the caller ID displayed on your CDR of your PBX server. A simplified HT503 peering scenario is presented on Figure 1.
Requirements:
  1. An HT503 with the latest firmware loaded.
  2. An Asterisk PBX Server ( this tutorial is using Elastix as a PBX)
  3. Prior to starting the configuration on the HT503, you should make sure that the caller ID is displayed on your analog phone without any intermediate device (PBX, ATA etc…). To do so, simply take your phone with a screen that is able to display the caller ID and plug it to the PSTN line. The caller ID is absolutely necessary if you want your server to log your calls and use information related to the caller ID. The phone should be able to display the caller ID to witness the outcomes of this tutorial.
HT502 peered with Asterisk/Elastix/Trixbox HT502 peered with Asterisk/Elastix/Trixbox
Figure 1: Simplified VoIP network in which the peering is made between the HT503 and Asterisk Server


Caller ID test on your HT503:
At first, we have to be sure that the HT503 handles the caller ID correctly; otherwise there will be no need to proceed to the next step. A verification to see if the HT503 handles the caller ID properly is needed.  You can verify if the caller ID is properly handled by doing like this:
  1. Open the web interface of your HT503.
  2. Click on the FXO port
  3. Scroll down of the page:
    1. Go to the Caller ID Scheme: Select the Caller ID Scheme that is supported by your phone service provider from the drop down menu
    2. Go to Caller ID Transport Type: Select the method by which the caller ID is transported.
    3. Scroll down of the page to Number of rings. This is the number of times the FXO port will be ringing before having the a VoIP extension ringing or before having the phone on the FXS port ringing if PSTN Ring through FXS is set to yes. Please set it to a value that is equal or greater than 4.
    4. Scroll down of the page to PSTN Ring Thru FXS to Yes, so the phone connected to the FXS port will ring.
    5. Click on update
    6. Check if the same Caller ID settings on the FXS port are similar to those on the FXO port.
    7. Go to the Basic Settings , and check if the Life Line Mode is set to Auto. Setting the Life Line Mode to Auto will allow the device to parse the Caller ID if the HT503 is on, and automatically be a relay if the device is off.
    8. Plug the phone to the FXS port, and do a PSTN call to the HT503.
    9. If you see the caller ID on your phone and the HT503 is on and all the previous parameters are respected, the HT503 handles properly the Caller ID. And now you  can move on to the next step.
Configuration of your HT503:
Now that you are sure that the HT503 is handling the Caller ID correctly, you should be able to do the peering of the HT503 with Elastix and having your CDR fetched with the Caller ID from the PSTN network and from VoIP to PSTN. As mentioned previously, the peering is without registration.
Please follow the steps below so you can configure your HT503 to route a call to an extension that you want to receive the call on. The settings below are either an addition or a change to the configuration that was made previously, so you can go through the Asterisk server to make the call to the FXS port of the HT503. Please make sure you go through all these settings:
Page Settings
Basic Settings Unconditional Call Forward: this option will allow you to forward a PSTN call to any extension that you want in the VoIP network. You can configure the HT503 to send the call to an extension that is registered in Elastix. Example: your Elastix PBX IP address is 192.168.1.254 and listening on incoming SIP connections on 5060, the FXS port of the HT503 is registered to Elastix and the extension number is 4001. So the Unconditional Call Forward will be 4001@192.168.1.254:5060
Advanced Settings Life Line Mode: as explained before this will change the behavior of the HT503 depending on whether it’s powered on or off. If the device is on, then It will handle the Caller ID and can transfer it to the Asterisk server, if not it will act just as a simple relay. But this time, it should not ring through the FXS port as the HT503 is set not to ring through it.




FXO Port
Account Active set to YES
Primary SIP server set to the IP address of the PBX
User ID is phone number set to NO
SIP Registration set to NO
Unregister on Reboot set to YES (this will allow the HT503 to clear all the SIP credentials that were set to be used on the device before and unregister from the server)
Outgoing Calls without Registration set to YES (this will allow to make PSTN call without being registered to a SIP server)
Number of Rings is greater than or equal 4
Stage Method set to 1
PSTN Ring through FXS set to NO
FXS Port Verify if the caller ID settings are the same as the ones defined on the FXO Port.
Register the FXS port on the Asterisk /Elastix server so you can use it with the phone
Configuration your Asterisk PBX (Elastix):
Please follow the steps below to configure your Asterisk IP PBX:
1 -Add A SIP Trunk from the Trunk men
2 -Under the outgoing settings:
A – Set a trunk name (i.e. HT503_Trunk)
B – For the PEER DETAILS enter the following:
  • host=IP_ADDRESS_OF_YOUR_HT503
  • type=peer
  • canreinvite=no
  • insecure=very
  • dtmfmode=rfc2833
  • nat=yes
  • port=5062 (this is the port number used by the HT503’s FXO port)
3 -Under the incoming settings:
A- Give the User Context to a certain name.
B-For the USER DETAILS enter the following:
  • context=from-trunk
  • host=dynamic
  • insecure=very
  • type=friend
  • dtmfmode=rfc2833
4 -Submit and Apply the settings.
5 -Go to the Outbound Routes:
A – Give the outbound route a name
B – Set the Dial Pattern if you want to dial a PSTN number.
C – Under Trunk Sequence, select the HT503 trunk that you have created before.
6 – Submit and apply the settings.
Final Result:
Now you should be able make and receive calls from and to a PSTN network, while having all the calls logged into the CDR of your server.
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7 comentarios:

  1. Muchas gracias amigo, me salvaste el día. Muy buena guía ! Es muy interesante que estos aparatitos HT503 de Grandstream puedan reemplazar a las costosas tarjetas de Digium o incluso Sangoma.

    Leí que Grandstream está auspiciando las capacitaciones de Elastix. Me parece una alianza excelente.

    Saludos.

    ResponderEliminar
  2. Recomiendo el uso de estos aparatos solo para un ambiente de laboratorio o uso personal. Si deseas una solución profesional , estable y de calidad te aconsejo que inviertas en productos Digium, Openvox, Sangoma. Otra cosa recuerda que los creadores de Asterisk fue Digium y comprando los productos Digium apoyas el crecimiento y el desarrollo de Asterisk del cual se beneficia tanto Elastix, como Grandstream

    ResponderEliminar
  3. Hello,

    Thank you for the tutorial, I am having trouble with incoming calls, the HT503 does not send the request to the sip Elastix.
    If I set the life line on then the FXS rings.
    I can dial out normally.
    On the sip server I do not seem to receive anything from the HT503 when an incoming call is happening. How can I debug the HT503 ? or what could be my problem ?
    Thank you in advance for trying to help.

    ResponderEliminar
  4. Sorry the problem was solved however it is more critical now.
    Incoming calls are only accepted if the "Allow sip guests = YES" under the
    Asterisk SIP settings. This render the PBX prone to easy attacks.
    Did you face that problem with your HT503 ? and what solution did you apply ?
    Thank you

    ResponderEliminar
  5. Hi.
    I've set the Allow sip guests=yes in the asterisk sip settings, but the incoming call still doesn't ring on the Unconditional Call Forward user ID, which is one of my extensions.
    Any tip?
    Best regards

    ResponderEliminar
  6. buenas, tengo el siguiente problema, configuro correctamente el ht503, pero cuando se cierra una llamada, en realidad asterisk la cierra alrededor de 40 o 50 segundos despues, he oido que eso es por no programar la "FRECUENCIA DE TONO DE DESCONEXION", alguien me puede ayudar en eso

    ResponderEliminar
    Respuestas
    1. Amigo tengo el mismo problema, e logrado encontrar casi todo los tonos de mi pais , solo me falta el de desconecxion te paso el link de la pagina donde puedes conseguirlos para tu pais : http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf. o tambien en la direccion : http://www.3amsystems.com/World_Tone_Database , espero que alguien me puede ayudar con el tono de desconecion de movistar en PERU!

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