Table of Contents

Using FreePBX with SDF VOIP

This tutorial shows how to add your SDF VOIP extension as a trunk in FreePBX.

Prerequisites

Procedure

In the admin interface follow these steps, replacing $EXTENSION and $PASSWORD with your extension number and password respectively:

  1. Go to Connectivity ยป Trunks
  2. Select Add Trunk and choose Add SIP (chan_pjsip) trunk*
  3. Set Trunk Name to sdf
  4. Set Outbound Caller ID to $EXTENSION@sip.sdf.org
  5. Set CID Options to Force Trunk CID
  6. Additional configuration for pjsip:
    1. Go to the pjsip Settings tab
      1. Under the General tab fill out the following
        1. Username: $EXTENSION
        2. Secret: $PASSWORD
        3. Authentication: Outbound
        4. Registration: Send
        5. SIP Server: sip.sdf.org
        6. SIP Port: 5060
        7. Context: from-trunk
        8. Transport: 0.0.0.0-udp
      2. Under the Advanced tab, most of the defaults should work okay, but make sure the following are set:
        1. DTMF Mode: Auto
        2. From Domain: sip.sdf.org
        3. From User: $EXTENSION
        4. Direct Media: No
    2. Click Submit and then Apply Config
  7. Additional configuration for sip (legacy):
    1. Go to the sip Settings tab
      1. Set Trunk Name to SDF
      2. Set peer details to:
        host=sip.sdf.org
username=$EXTENSION
secret=$PASSWORD
type=friend
context=from-trunk
insecure=port,invite
trustrpid=yes
sendrpid=yes
directmedia=no
qualify=yes
keepalive=45
nat=yes
dtmfmode=rfc2833
disallow=all
allow=ulaw
      3. Go to the Incoming tab
      4. Set Register String to $EXTENSION:$PASSWORD@sip.sdf.org
      5. Click Submit

You can now define Inbound and Outbound routes using this new trunk. You may refer to FreePBX documentation on how to set up outbound and inbound routes. To test inbound calls, you can queue a test call through maint.

* NOTE: Older versions of FreePBX may still allow for chan_sip trunk settings, but this is deprecated and chan_pjsip is now the preferred configuration.

If you have an incoming DID through SDF, just a note that the SDF's system does not pass FROM_DID=.
If you would like to route your call through a specific incoming route by DID, then add the DID after the registration string.
For example: $EXTENSION:$PASSWORD@sip.sdf.org/$DID You can add the 11 or 10 digit number, just make sure it matches exactly to what you want in the incoming route.

$Id: sdf_voip_freepbx.html,v 1.2 2017/11/07 12:49:54 irl Exp $ Using FreePBX with SDF VOIP - traditional link (using RCS)