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    New FreePBX Installation

    IT Discussion
    freepbx freepbx 14 freepbx setup twilio asterisk
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    • A
      Alex Sage
      last edited by Alex Sage

      So I setup a fresh FreePBX 14 system.

      Got Zoiper setup on Linux and iOS - Check

      I am using Twilio so I got outgoing calling working - Check.

      Incoming calling would not work...

      Found this: http://hwdevelopment.com/blog/27-freepbx-13-asterisk-11-twilio-elastic-sip-trunk-setup

      So I change the settings as referenced above and Twilio started to work for incoming calls.

      However both Zoiper clients stopped working 😞

      So should I be using chan_pjsip or chan_sip for trunks? What about Phones, and Softphones? Both? On what port?

      EddieJenningsE Emad RE 2 Replies Last reply Reply Quote 0
      • EddieJenningsE
        EddieJennings @Alex Sage
        last edited by

        @aaronstuder Twilio works with PJSIP just fine. No need to do special stuff to use ChanSip.

        A 1 Reply Last reply Reply Quote 0
        • A
          Alex Sage @EddieJennings
          last edited by Alex Sage

          @eddiejennings care to share your setup?

          EddieJenningsE 1 Reply Last reply Reply Quote 0
          • EddieJenningsE
            EddieJennings @Alex Sage
            last edited by

            @aaronstuder In the middle of a game. I can when I get to a stopping point.

            A 1 Reply Last reply Reply Quote 0
            • A
              Alex Sage @EddieJennings
              last edited by

              @eddiejennings Thanks!!!

              1 Reply Last reply Reply Quote 0
              • EddieJenningsE
                EddieJennings
                last edited by

                Nothing special.

                1. I did not change any of the default port settings on FreePBX, so pjsip is bound to 5060.
                2. In the Twilio settings you'll need to set your origination URI to sip:yourhost.domain.tld or sip:ipAddress
                3. In FreePBX, you'll setup a PJSIP trunk using the credentials you created in authentication section of Elastic SIP trunking.
                4. Make sure you use one of the addresses from your termination URIs.
                5. In pjsip advanced settings set expiration to 120 (like what voip.ms says)

                I, like Jared, do my dialing patterns on outbound routes rather than on trunks. You said you had termination working, so you've got the +1 setup correct.

                A 1 Reply Last reply Reply Quote 1
                • A
                  Alex Sage @EddieJennings
                  last edited by

                  @eddiejennings what version of FreeFBX? Asterisk?

                  EddieJenningsE 1 Reply Last reply Reply Quote 0
                  • EddieJenningsE
                    EddieJennings @Alex Sage
                    last edited by

                    @aaronstuder FreePBX 14 and Asterisk 13 I believe. Whatever the recommended thing was when I installed it.

                    A 1 Reply Last reply Reply Quote 0
                    • A
                      Alex Sage @EddieJennings
                      last edited by

                      @eddiejennings Thanks! I’ll have go try again in the morning.

                      EddieJenningsE 1 Reply Last reply Reply Quote 0
                      • EddieJenningsE
                        EddieJennings @Alex Sage
                        last edited by

                        @aaronstuder Assuming I have a stable Internet connection, I'll be checking ML throughout the day, so if you have problems, post here. 😄

                        A 1 Reply Last reply Reply Quote 0
                        • A
                          Alex Sage @EddieJennings
                          last edited by

                          @eddiejennings Will do, Thanks

                          1 Reply Last reply Reply Quote 0
                          • Emad RE
                            Emad R @Alex Sage
                            last edited by Emad R

                            @aaronstuder said in New FreePBX Installation:

                            So I setup a fresh FreePBX 14 system.

                            Got Zoiper setup on Linux and iOS - Check

                            I am using Twilio so I got outgoing calling working - Check.

                            Incoming calling would not work...

                            Found this: http://hwdevelopment.com/blog/27-freepbx-13-asterisk-11-twilio-elastic-sip-trunk-setup

                            So I change the settings as referenced above and Twilio started to work for incoming calls.

                            However both Zoiper clients stopped working 😞

                            So should I be using chan_pjsip or chan_sip for trunks? What about Phones, and Softphones? Both? On what port?

                            So should I be using chan_pjsip or chan_sip for trunks? What about Phones, and Softphones? Both? On what port?

                            I used AIX for everything, and didnt come close to any setting for SIP.

                            Just created AIX extension and used port 4569 I even changed that in the extension page to another port and it works, calls from LAN and calls from WAN.

                            In Zoiper there is an option to configure an IAX account, the reason I choose IAX cause it does everything in 1 port and options seemed more easier than SIP

                            But video didnt work, only audio + call recording .

                            EddieJenningsE 1 Reply Last reply Reply Quote 0
                            • EddieJenningsE
                              EddieJennings @Emad R
                              last edited by

                              @emad-r in my limited experience in VoIP, I've only used PJSIP. I don't know enough about IAX to give you an assessment of the two.

                              1 Reply Last reply Reply Quote 0
                              • A
                                Alex Sage
                                last edited by

                                Trying again now...

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

                                  @Emad-R It is IAX2 not AIX.

                                  IAX2 is Inter-Asterisk Exchange and is not a standard for everything. Very little support can be found for IAX2 in general.

                                  IAX2 is nice because it only uses one port (UDP 4569) for signaling and audio, but again, not a heavily used protocol.

                                  The SIP protocol as implemented in Asterisk was showing its age and limiting options in the modern world. That is why Asterisk developers came up with PJSIP, to extend funcitonality while also retaining basic compatibility.

                                  Asterisk (not FreePBX) recommends that you should be using PJSIP by default now.

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

                                    @aaronstuder there is no significant difference between using SIP or PJSIP for a trunk connection. PJSIP attempts different signaling, but will still signal to match SIP if needed.

                                    A 1 Reply Last reply Reply Quote 0
                                    • A
                                      Alex Sage @JaredBusch
                                      last edited by

                                      I pulled a @jaredbusch and justed used voip.ms 😄

                                      1 Reply Last reply Reply Quote 0
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