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    Setting up a SIP trunk in FreePBX 13

    MangoCon
    freepbx freepbx 13 freepbx setup sip trunk voip.ms guide real instructions how to jareds guide to freepbx 13
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    • JaredBuschJ
      JaredBusch
      last edited by

      Log in to VoIP.ms and navigate to DID Numbers -> Manage DID(s)
      0_1476638384130_upload-95d4c758-9855-4864-ac61-71f09937efec

      Look for the DID you want to use for the trunk and note the number, routing, and POP.
      0_1476640236234_upload-3fd76fa1-0b52-4475-adb2-6799ac6d8b09

      In FreePBX, navigate to Connectivity -> Trunks
      0_1476640438225_upload-d3a447f9-d56f-4fb2-aaa8-9fe63b0c36b5

      Click +Add Trunk -> +Add SIP (chan_pjsip) Trunk. PJSIP simplifies the setup from the PBX side and is the new default for Asterisk. You can read all about it straight from Digium if you want.
      0_1476640490338_upload-32f5d475-c899-4d90-b8bc-a32c700fd152

      Fill out the General tab as desired. Note the outbound Caller ID format and also set a reasonable maximum channels value to protect against extreme charges if a SIP user's credentials are ever compromised. I generally leave the Caller ID blank and set it on the outbound routes, but it is easy to miss there.
      0_1476640752208_upload-0b156c68-ca0e-4795-b0a7-95fdb58f08f8

      Leave the Dialed Number Manipulation Rules empty. I always recommend handling that on the outbound route(s).
      0_1476640911883_upload-7a494eb4-5f6d-4fde-8faf-b61613554946

      Fill out the General tab on PJSIP Settings with your voip.ms information noted from above. The username is the account number, or if you use a sub account it is the account number _subaccountname. The secret is the password for the account on VoIP.ms. The SIP server is the pop name with the dash removed and .voip.ms appended.
      0_1476641322364_upload-dbe1d28f-1de6-4e8b-b2fd-0acb7ff7751a

      Click to the advanced tab
      0_1476641382336_upload-ca5defef-24ee-46f4-b669-9c2cfb61a7bd

      Scroll to the bottom and change detect fax to enabled if you want to enable inbound faxing on this trunk. VoIP.ms does not support T.38 FoIP but faxing over SIP works for most simple needs.
      0_1476641502737_upload-055e48d8-e122-41ff-b2a2-808da3a6e38e

      You can leave the codecs tab alone unless you have changed default codecs on VoIP.ms side.
      0_1476641654210_upload-4e92834d-1dd8-4286-bcba-2f247175f28b

      Click submit at the bottom of the screen, and then after it finishes, click the red Apply Config button.
      0_1476641702807_upload-5b6306cf-871e-45e9-8758-6efe383600bf

      Go back to the FreePBX dashboard and you should see the trunk online after you click the little circle arrows to refresh.
      0_1476641872530_upload-6d63c9a1-8a9b-4b04-9fb2-4103a46371ee

      Part of the FreePBX 13 Setup Guide

      1 Reply Last reply Reply Quote 2
      • JaredBuschJ
        JaredBusch
        last edited by JaredBusch

        Some people have reported issues using the PJSIP settings with VoIP.ms. I have been using it for months with no problems, but if you are having problems, then you can create a CHAN_SIP based trunk with no problems.

        If you have problems with PJSIP, or jsut need to confirmed support of CHAN_SIP from your provider, setting up a CHAN_SIP trunk is not much more difficult.

        Choose CHAN_SIP
        0_1519669532602_c270fc33-d10b-4038-a595-ba27f603cd53-image.png

        The General tab gets filled out the same as with PJSIP.
        0_1519669635788_4b349bcf-9483-4b76-9566-a1cc468bdb6a-image.png

        Leave the Dialed Number Manipulation Rules empty. I always recommend handling that on the outbound route(s).
        img

        Remove the default information in the Outgoing tab
        0_1519669770315_2df779fe-30fe-4697-a026-c37924e7cbff-image.png

        And populate it with your information. No spaces or characters in the trunk name as that gets used as the Asterisk context.
        0_1519669863014_92f24815-b398-4cd1-8389-880501c89364-image.png

        Here are the settings, you should only need to edit the first 4 lines appropriately.

        host=chicago2.voip.ms
        username=123456_some_sub_acct
        fromuser=123456_some_sub_acct
        secret=some_password
        type=peer
        context=from-trunk
        trustrpid=yes
        sendrpid=yes
        qualify=yes
        nat=yes
        directmedia=no
        insecure=port,invite
        disallow=all
        allow=ulaw
        

        On the Incoming tab , again delete the default data.
        0_1519669984732_7d8ad621-b2a7-41b2-8efd-6a0e826e446d-image.png

        This time leave the USER Context and USER Details blank and only populate the registration string. This is why you want to avoid @ or : in your SIP trunk password.
        0_1519670026551_717bc2cb-0d99-41cb-8380-3da657936978-image.png

        1 Reply Last reply Reply Quote 0
        • scottalanmillerS
          scottalanmiller
          last edited by

          Just followed this, worked great.

          1 Reply Last reply Reply Quote 0
          • JaredBuschJ
            JaredBusch
            last edited by JaredBusch

            Following up on this way too many weeks later...

            Sorry about that...

            With VoIP.ms and PJSIP, you also need to go to the advanced tab and change the default expiration time from 3600 to 120.

            I have not tested this with other providers yet as it would be disruptive to a client.
            When I have time I will setup a backup provider and change my extension to use that normally and see how things go. The problem is that with a not very used system, it will still reregister every hour, so to catch it, I have to make a call between the fial time and the reregister time.

            0_1498692886260_28f1a3e4-688d-45bc-8747-11cf87efbca2-image.png

            1 Reply Last reply Reply Quote 1
            • JaredBuschJ
              JaredBusch
              last edited by

              Post 2 updated with the CHAN_SIP instructions.. sorry to be so bad about this.

              https://mangolassi.it/topic/12327/setting-up-a-sip-trunk-in-freepbx-13/2

              1 Reply Last reply Reply Quote 0
              • u2communicationsU
                u2communications
                last edited by

                If you find yourself still trying to get incoming calls working after several hours (like me), be advised that the default DID settings on voip.ms will not work. You merely have to set up a sub-account and make sure to set the authentication type as user/password authentication. Add your DID's to the sub-account and authenticate with the accountnumber_subaccount convention and use the password you created when setting up the sub-account.

                JaredBuschJ 1 Reply Last reply Reply Quote 2
                • JaredBuschJ
                  JaredBusch
                  last edited by

                  Follow up, You also need to set max retries to 0 or the PJSIP trunk can potentially go offline and stop trying.

                  1 Reply Last reply Reply Quote 0
                  • JaredBuschJ
                    JaredBusch @u2communications
                    last edited by JaredBusch

                    @u2communications said in Setting up a SIP trunk in FreePBX 13:

                    If you find yourself still trying to get incoming calls working after several hours (like me), be advised that the default DID settings on voip.ms will not work. You merely have to set up a sub-account and make sure to set the authentication type as user/password authentication. Add your DID's to the sub-account and authenticate with the accountnumber_subaccount convention and use the password you created when setting up the sub-account.

                    I have never experienced an issue using the primary account on VoIP.ms. I highly recommend against it, always. But, I have never had a problem using it.

                    Also, welcome to ML.

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