FreePBX – Basic Configuration Guide

asteriskfreepbxsip

I've just successfully set up an Asterisk+FreePBX install on an Amazon EC2 instance per this guide: http://voxilla.com/2009/10/15/voxillas-freepbx-in-a-cloud-step-by-step-1457 (I've also assigned it an Elastic IP).

I'd love to test it without buying any VoIP trunks yet. I understand that I can do that by just placing internal calls to two phones connected to the Asterisk server, right?

I'm using a softphone: CounterPath eyeBeam.

I can't seem to find any guide or resource online that will tell me what I need to do on the FreePBX side of things to set it up for just routing internal calls, nor what I need to enter in to CP's eyeBeam client (username, auth. name, password, etc.) in order to get it to connect to Asterisk.

Can somebody give me a quick primer? Or at least point me to an easily-digestible resource that could help me out?

EDIT: I did already find this guide, but it sets up the FreePBX to use outside VoIP trunks, which I don't want to purchase just yet.

Best Answer

It is enough to just add two SIP extensions, Username, display name, password, apply changes, and in your eyeBeam add username, password and the IP address of your Asterisk as Domain (I don't have it installed, the domain part may be called differently). Yes, it is that simple.

Related Solutions

Nat – asterisk/freeswitch in nat/no-nat setup

For FreeSWITCH, I believe this functionality can be handled by the internal/external sip profiles. You can find more info on wiki.freeswitch.org

The NAT configuration to your external VoIP provider(s) can be setup in the external profile (example /usr/local/freeswitch/conf/sip_profiles/external/voipprovider.xml). You can set these two parameters to match your public ip address:

<include>
        <gateway name="voipprovider">
        <param name="ext-rtp-ip" value="x.x.x.x"/>
        <param name="ext-sip-ip" value="x.x.x.x"/>
        <param name="username" value="user"/>
        <param name="password" value="password"/>
        <param name="realm" value="sip.voipprovider.com"/>
        </gateway>
</include>

To use your external voip providers, you would setup your dialplan (/usr/local/freeswitch/conf/dialplan/default.xml) something like this. To use different voip providers depending on destination, you can adjust the regex:

<include>
 <extension name="Dial Out VoIP Provider">
   <condition field="destination_number" expression="^9(1\d{10})$">
     <action application="bridge" data="sofia/gateway/voipprovider/$1"/>
   </condition>
 </extension>
</include>

I hope this helps. I don't have any experience w/ Asterisk, so I am not sure how you would set it up with that switch.

Automated reports with FreePBX 2.8.1.4

Shameless plug :) but you could try my product SAMReports, it works with FreePBX, as well as with other Asterisk editions .

Best Answer

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Nat – asterisk/freeswitch in nat/no-nat setup

Automated reports with FreePBX 2.8.1.4

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