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By WelshPaul
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Below is my Sipgate Basic PJSIP configuration that I use with my FreePBX setup. So far so good! :nerd:

First thing you will need to do is enable the "SIP Channel Driver" to use both chan_sip and chan_pjsip. You can do that by navigating to the Settings > Advanced Settings configuration page and scrolling down until you see the "SIP Channel Driver" setting. Make sure that "both" is selected in the dropdown box. If it isn't, make the change and click "Submit" to save it and then "Apply Changes" to implement it.

Now that's out of the way, let's create a Sipgate Basic PJSIP Trunk...
  1. Navigate to Connectivity > Trunks > + Add Trunk > Add SIP (chan_pjsip) Trunk page.
    You should be at the following screen:
  2. Under the "General" tab section make the following changes:
    • Trunk Name = Sipgate-(SIP-ID)
    • Outbound CallerID = (PHONE-NUMBER)
    It should look something like this:
  3. Click on the "pjsip Settings" tab and in the "General" tab section make the following changes:
    • Username = (SIP-ID)
    • Secret = (SIP-PASSWORD)
    • Authentication = Outbound
    • Registration = Send
    • Language Code = English - United Kingdom
    • SIP Server =
    • SIP Server Port = 5060
    • Context = from-trunk
    • Transport =
    Again, it should look like this:
  4. While still in the "pjsip Settings" tab click on the "Advanced" tab and make the following changes:
    • Forbidden Retry Interval = 20
    • Fatal Retry Interval = 20
    • General Retry Interval = 20
    • Expiration = 600
    • Qualify Frequency = 20
    • Contact User = (SIP-ID)
    • From Domain = (YOUR-IP)
    • From User = (SIP-ID)
    • Client URI = sip:(SIP-ID)
    • Server URI =
    • AOR Contact = sip:(SIP-ID)
    • Match (Permit) =
    You can leave the rest at their defaults.
  5. Finally click on the "Submit" button to save your changes and then "Apply Changes" to implement them.
Additional information on how to convert extensions from chan_sip to chan_pjsip can be read here: WelshPaul @ FREEPBX - New tool to assist converting from SIP to PJSIP

NOTE: Replace the following (in the above configuration) with your real Voipfone credentials:
  1. (SIP-ID) = Your Sipgate Basic SIP-ID number.
  2. (PHONE-NUMBER) = Your Sipgate Basic Phone Number
  3. (SIP-PASSWORD) = Your Sipgate Basic SIP Password.
  4. (YOUR-IP) = Your IP Address (Find it here: Settings > Asterisk SIP Settings > NAT Settings > External Address)
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