Trixbox™ - open platform for business telephony
Why use trixbox CE?
trixbox Community Edition began in 2004 as the massively popular Open Source IP-PBX project named Asterisk@Home. Since then, it has grown into the world's most popular distribution of Asterisk with over 65,000 downloads per month. trixbox is known for its flexibility to satisfy the needs of custom deployments and continues to be FREE .
trixbox CE allows you to build your own custom features and modules. The trixbox community is one of the largest and most active communities of trixbox and Asterisk users in the world. The members of this community work every day to help each other answer questions, resolve issues, fix bugs, make enhancements, and develop projects. trixbox CE has all the benefits of open source plus a commercial company standing behind it.
Configuring trixbox trunk for inbound calls
Connect to your trixbox using a PC in your network by typing HTTP://YourAsteriskIpaddress in your web browser.
Select FREEPBX under the Asterisk Menu
Click Trunks then - Add SIP Trunk.
Enter the following information (see example below):
Click Submit Changes.
Next go to the “Inbound routes” Tab to set the call routings. Populate “DID Number” field with a particular DID that you wish to route to a destination or alternatively, you can leave it empty, and all DIDs will be routed to one selected route.
Click Submit to apply changes.
STEP 8 (OPTIONAL)
This step is optional, and may only be needed if your trixbox server has a private IP Address. This step may also be needed if you notice one-way audio.
Open the SIP_NAT.CONF file.
Enter the following information at the top of your sip.conf file in the general section of the SIP_NAT.CONF file:
externip=WAN IP address
Now try calling your DID. The destination you choose should ring or answer.
Note: If you notice one-way audio, see step 8 above.