*****NOTE*****This document is deprecated. Please see OnSIP Trunking. 07/30/14

Below are the configuration Instructions for the FreePBX portion of the Trixbox.

OnSIP recommends creating both a SIP and IAX trunk. Then, you have flexibility and redundancy in your dial plan.

Create a SIP Trunk

The first thing to do is to log into your Trixbox. You should log in a user "maint."

Now select the FreePBX selection under the Asterisk menu option.

From here choose Setup from the menus in the middle of the screen.

From the "Setup" menu choose Trunks and then choose Add SIP Trunk. Create a SIP Trunk.

Outgoing Settings

Trunk Name = junction
context=from-pstn
fromdomain=sip.jnctn.net
fromuser=MY_USERNAME
host=sip.jnctn.net
insecure=invite 
username=AUTH_USERNAME
secret=MY_PASSWORD
type=peer

Incoming Settings

Leave Incoming Settings blank for SIP
Registration String: 
register = MY_USERNAME@sip.jnctn.net:MY_PASSWORD@sip.jnctn.net

Click Submit. Your SIP trunk is now complete. On to your IAX trunk.

Create a IAX Trunk

Outgoing Calls

Now select the Asterisk Management Portal. You should see the following screen:

From here choose Setup from the menus in the middle of the screen.

From the "Setup" menu choose Trunks and then choose Add IAX2 Trunk.

1.) Caller ID: Under General Setting put in your Caller ID. The format is:

"Name" <phone number>

No information is needed under "Outgoing Dialing Rules".

2.) Outgoing Settings:

Enter the following in the Outgoing Settings box:

host=iax.jnctn.net
secret=your_password
type=peer
username=your_username

Be sure to replace "your_password" and "your_username" is your VOIP password and username from OnSIP. Name the trunk "junction".

3.) At the bottom of the page choose Submit Changes.

4.) From the "Setup" menu, choose Outbound Routes. Make the "junction" trunk your sequence "0" (zero) trunk under the "Outside" route.

**
Important - We are expecting a "1" before the number.
When dialing, be sure to dial 1-Nxx-Nxx-xxxx.
**

Incoming Calls

Follow the instructions for Outgoing Calls. Make sure you can make calls.

1.) We use RSA keys for authentication purposes. You will need to SSH to your box and go to a command line. This step is very important. Without our RSA keys, you will not be able to receive calls.

From the command prompt enter the following:

cd /var/lib/asterisk/keys
wget http://www.jnctn.com/jnctn.pub

2.) Incoming Settings:

From the Setup menu, choose Setup. Choose the Trunk IAX2/junction trunk.
Enter the following in the Incoming Settings box:

auth=rsa
context=from-pstn
inkeys=jnctn
type=user

Name the trunk "jnctn".

***
For Trixbox users using SIP protocol, leave the Incoming Settings box and name completely blank.
***

3.) Registration:

Enter the follwoing into the Registration box:

your_username:your_password@iax.jnctn.net

Be sure to replace "your_password" and "your_username" is your VOIP password and username from OnSIP. From the bottom of the page choose Submit Changes.

4.) Setup DID:

In the "Setup" menu, choose Inbound Routes. Choose Add DID. Enter the phone number you were given by Junction Networks. Be sure to add all 11 digits plus the "1". For example, if your DID is 212-555-1234, you would enter:

12125551234

Congratulations. You should now be able to make and receive calls from OnSIP!