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By WelshPaul
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Before we start, you should understand that A&A’s offer two different methods we can use to register a trunk against:
  1. SIP Phone
  2. To your server via SIP
I chose the “SIP Phone” option because I have an Andrews & Arnold SIP2SIM registered to an extension on the same PBX which fails to register when using the “To your server via SIP” option. Don't worry if you don't have or use Andrews & Arnold's SIP2SIM service, this configuration will work for both users of the SIP2SIM service and users who don't.

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 an Andrews & Arnold 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 = AAISP-(441234567890)
    • 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 = (ACCOUNT)
    • Secret = (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:
    • DTMF Mode = RFC 4733
    • Forbidden Retry Interval = 20
    • Fatal Retry Interval = 20
    • General Retry Interval = 20
    • Expiration = 60
    • Qualify Frequency = 20
    • Contact User = (ACCOUNT)
    • From Domain = (YOUR_IP)
    • From User = (ACCOUNT)
    • Client URI = sip:(ACCOUNT)
    • Server URI =
    • AOR Contact = sip:(ACCOUNT)
    • Match (Permit) =
    You can leave the rest at their defaults. It should look like this:
  5. Click on the "Codec" tab and select the following codecs:
    • alaw
    • ulaw
    Again, it should look like this:
  6. 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 Andrews & Arnold credentials:
  1. (ACCOUNT) = Your Andrews & Arnold phone number in international format (+441234567890).
  2. (PASSWORD) = Your Andrews & Arnold SIP Password. Your SIP Password can be found in the Andrews & Arnold Control Pages under the "Outgoing" tab.
  3. (YOUR_IP) = Your IP Address (Find it here: Settings > Asterisk SIP Settings > NAT Settings > External Address)
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By WelshPaul
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Below is my Dial Number Manipulation Rules. You can use mine or you can use your own custom rules! If you use mine, make sure that you replace 01792 with your own area code.

Dial Number Manipulation Rules:
  • Prepend: 01792 Prefix: match pattern: NXXXXX
  • Prepend: 14101792 Prefix: 141 match pattern: NXXXXX
  • Prepend: 147001792 Prefix: 1470 match pattern: NXXXXX
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By jamesg225
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Thank you for the information. I have my outgoing calls working fine but incoming are not working. Other trunks with Voipfone and Sipgate are working fine - I'm behind NAT and don't particularly want to open up the firewall. Could I be missing something?

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