Connecting a SIP Trunk to FreePBX / Asterisk

Basic Sip trunk configuration for adding VoiceCloud as a SIP Trunk to your Asterisk-based PBX

Create SIP Trunk

1. Using Chrome or Firefox navigate to the web console of the PBX.
2. Click on FreePBX Administration.
   
3. Log in with your administrator credentials.
4. From the navigation at the top select Connectivity and then Trunks.
   
5. Click on + Add Trunk and then + Add (chan_sip) Trunk.
   
6. Trunk Name: Hosted PBX
   
7.  Click on the tab for sip Settings.
   
8. On the tab for Outgoing fill out the following details
   • Trunk Name: pbx-out
   • PEER Details:

     username={YOURSIPUSERNAME} // e.g  '5001-USERNAME'
     type=friend
     trustrpid=yes
     sendrpid=yes
     secret={YOURSIPPASSWORD} // e.g  '*************'
     qualify=yes
     nat=yes
     keepalive=60
     insecure=port,invite
     host=sip1.giant.net.uk
     fromuser={YOURSIPUSERNAME} // e.g  '5001-USERNAME'
     fromdomain=sip.giant.net.uk
     dtmfmode=inband
     relaxdtmf=yes
     disallow=all
     directmedia=no
     context=from-trunk
     allow=ulaw // Any other codecs you may need

   • Click Submit.
     

9. You will will receiving a popup. Click OK
   

10. Click Apply Configuration.