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Connecting a SIP Trunk to FreePBX / Asterisk

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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.
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  3. Log in with your administrator credentials.
  4. From the navigation at the top select Connectivity and then Trunks.
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  5. Click on + Add Trunk and then + Add (chan_sip) Trunk.
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  6. Trunk Name: Hosted PBX
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  7.  Click on the tab for sip Settings.
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  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.
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  9. You will will receiving a popup. Click OK
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  10. Click Apply Configuration.
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