====== Using FreePBX with SDF VOIP ======

This tutorial shows how to add your SDF VOIP extension as a trunk in [[https://www.freepbx.org/|FreePBX]].

===== Prerequisites =====

  * A working FreePBX setup
  * Access to the FreePBX admin interface

===== Procedure =====

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

  - Go to //Connectivity// » //Trunks//
  - Select //Add Trunk// and choose //Add SIP (chan_pjsip) trunk//*
  - Set //Trunk Name// to ''sdf''
  - Set //Outbound Caller ID// to ''<$EXTENSION@sip.sdf.org>''
  - Set //CID Options// to //Force Trunk CID//
  - Additional configuration for pjsip:
    - Go to the //pjsip Settings// tab
      - Under the //General// tab fill out the following
        - Username: ''$EXTENSION''
        - Secret: ''$PASSWORD''
        - Authentication: Outbound
        - Registration: Send
        - SIP Server: ''sip.sdf.org''
        - SIP Port: ''5060''
        - Context: ''from-trunk''
        - Transport: ''0.0.0.0-udp''
      - Under the //Advanced// tab, most of the defaults should work okay, but make sure the following are set:
        - DTMF Mode: Auto
        - From Domain: ''sip.sdf.org''
        - From User: ''$EXTENSION''
        - Direct Media: No
     - Click //Submit// and then //Apply Config// 
  - Additional configuration for sip (//legacy//):
     - Go to the //sip Settings// tab
       - Set //Trunk Name// to ''SDF''
       - Set peer details to:<code>
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
</code>
       - Go to the //Incoming// tab
       - Set //Register String// to ''$EXTENSION:$PASSWORD@sip.sdf.org''
       - 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 $ [[http://sdf.org/?tutorials/sdf_voip_freepbx|Using FreePBX with SDF VOIP]] - traditional link (using [[wp>Revision_Control_System|RCS]])