Posts Tagged ‘FreePBX’

Receive and place wireless or 3G calls from any Android based smartphone with 3CXPhone for Android

Friday, April 22nd, 2011

I have created a telephone extension for my home Asterisk/FreePBX server on my smartphone. This means I can now answer and make calls from my home phone lines when I am away from home as long as I have a Wi-Fi or 3G signal.  Here are the details from the 3CX website:

With 3CXPhone for Android, users can make free phone calls via wireless and 3G (mobile operator permitting). Examples of Android based phones are the HTC Desire, HTC Hero, Sony Ericsson Xperia, X10 Mini and Pro, Samsung Galaxy S and the Google Nexus.3CXPhone for Android works with standards based SIP servers – including 3CX Phone System, Asterisk and popular SIP VoIP Providers. Unlike other “free” Android SIP phones, 3CXPhone is NOT locked down to a particular provider or PBX.

Users can pick and choose their favorite provider or VoIP PBX and switch at any time or use different VoIP providers simultaneously. 3CXPhone is available free of charge for commercial and non profit organizations as well as private users.

“Smart phones will soon be the defacto mobile phone and with a docking station will act as a desk phone in time. A VoIP PBX must embrace this technology and deliver seamless integration to provide true mobility to its users.” said Nick Galea, CEO, 3CX.  “Smartphone support is a key component of our strategy and the availability of 3CXPhone for Android delivers on this vision”. Nick Galea continued “We chose Android as our first platform – it is rapidly gaining market share because it’s open, standards based, vendor independent and evolving at a rapid pace. Android based Smartphones are available in different form factors and at competitive prices.”3CXPhone for Android is based on SipAgent, a popular SIP phone for Android that 3CX acquired in June 2010. SipAgent users are able to upgrade free of charge to 3CXPhone.

3CXPhone for Android 3CXPhone for Android 3CXPhone for Android

Features include:

  • Fully supports the industry leading SIP standard
  • G711 and GSM codec support
  • Ability to transfer calls
  • Choose between integrated or custom dial pad
  • Record phone calls
  • Force VoIP calls when available – with this feature you can automatically make free VoIP calls take priority over 3G
  • Auto provisioning – SIP PBX’s or VoIP providers can automatically provision 3CXPhone for Android via multicast or HTTP

3CXPhone for Android is available for download here.

Earn More Money! Click here for info or contact Patrick Markham

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How to configure Portech MV-370 GSM with an Asterisk FreePBX Trunk

Friday, April 15th, 2011

Portech MV-370This guide will help you set up a Trunk in Asterisk (FreePBX, Trixbox, PBIF, etc.) for the Portech GSM Gateway.

This configuration will pass Caller ID.

First we will configure the Portech MV-370.   This configuration will also work with Portech MV-372 and other Portech MV-3xx like MV-374.

Login to your Portech

Route

  • Mobile To Lan Settings:
    Item CID URL
    0 * 192.168.x.x (your asterisk ip)
  • Lan To Mobile Settings:
    Item URL Call num
    0 * #
  • Mobile
    • Settings:
      Mobile 1:
      Sip From: Tel/Tel (Not reg)
      CLID Presentation: On
      LAN Answer Mode: Income
  • SIP Settings
    • Service Domain
      You only fill Domain Server and Proxy Server with your Asterisk IP address:
      Domain Server: 192.168.x.x
      Proxy Server: 192.168.x.x

    Again do the same for Mobile  2 if you are using the MV-372

    • Port Settings
      Make sure SIP Port for Mobile 1 is 5060 (and port 5062 for Mobile 2) 

    Don’t forget to save changes (should reboot after saving)

    Asterisk/FreePBX

    Login to your FreePBX and add SIP Trunk

    Outbound Called ID: 0711111111 (put your GSM number here)
    Maximum Channels: 1

    Outgoing settings
    Trunk Name: SIM1 (you may put anything you like)

    PEER Details:
    host=192.168.x.x (your Portech IP address)
    type=peer
    port=5060

    Incoming Settings:
    USER Context: 07111111111 (Your GSM mobile number)
    Leave Incoming Settings blank.

    Click submit (don’t forget the Orange bar on top after you make changes)

    If you have the dual-sim MV-372, add another SIP Trunk for SIM2.

    Outbound Called ID: 0722222222 (put your second GSM phone number here)
    Maximum Channels: 1
    go to Outgoing settings

    Trunk Name: SIM2
    PEER Details:
    host=192.168.x.x (your Portech IP address)
    type=peer
    port=5062 (important)

    Incoming Settings:
    USER Context: 0722222222 (put your second GSM phone number)

    Again apply changes.

    In General Settings, enable Allow Anonymous Inbound SIP Calls

    Now to make this work we have to create Outbound Route, so click Outbound Routes
    Put Route name as you wish, I have called it GSM_1 (then you can add another and make it GSM_2)

    Dial Patterns:  I have put 07xxxxxxxxx because I want only mobile numbers  to go through this trunk  (Portech). It’s all about good cost managment.

     Also don’t forget in order to receive calls you need to have Inbound Route setup on Asterisk/FreePBX. Create a new Inbound Route.  Set you destination to an extension or ring group or any other destionation you would like to transfer calls to. 

    I have set the CID Lookup source to ‘Phonebook’ so that I can see who is calling.  This is a feature that would not work when I had the Portech coming straight in on a Ring Group.

    Only thing you left to do now is click Submit Changes then Apply Configuration Changes and pray for this to work!

    CREDIT: http://xtittle.blogspot.com/2009/02/updated-configuration-asterisk-freepbx.html

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