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    Hosted VOIP and pfSense

    Scheduled Pinned Locked Moved NAT
    25 Posts 7 Posters 16.6k Views
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    • M
      mst
      last edited by

      localnet=192.168.1.0/255.255.255.0  make it to match your network

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      • R
        rugby
        last edited by

        @mst:

        localnet=192.168.1.0/255.255.255.0  make it to match your network

        Our PIAf hosted box has a public IP, do I still need this?  Our setup worked perfectly fine with an SG565 in place and Sip Proxy turned on.

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        • M
          mst
          last edited by

          if you have public IP then no

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          • R
            rugby
            last edited by

            @mst:

            if you have public IP then no

            Thanks for the clarification.  I didn't think it was needed.  Our phones just unregistered again.  I'm pulling this box until this issue is fixed somehow.  I'm beyond frustrated and we NEED our IP Phones to work reliably.

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            • M
              mst
              last edited by

              check this post:  http://www.trixbox.org/forums/vendor-moderated-forums/aastra-endpoints/57i-not-registering-no-service  can be usefull

              MST

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              • R
                rugby
                last edited by

                My phone ARE getting separate ports when they boot up initially, they only lose the registration when they try and re-register.  I put the SG565 back into service at that office and the phones have been rock solid for the past few hours.

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                • S
                  Supermule Banned
                  last edited by

                  I use Askozia PBX in VmWare setup… Works like a charm.....

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                  • R
                    rugby
                    last edited by

                    @Supermule:

                    I use Askozia PBX in VmWare setup… Works like a charm.....

                    I don't think this has to do with the PBX so much as the natting of SIP ports.  We are going to demo OnSIP in the coming weeks and I saw one of the threads pertaining to SIP nat.

                    I'm just frustrated because this should just work and it's "sort of" working which is worse than not working at all.

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                    • R
                      rugby
                      last edited by

                      I tweeted about my problems and Chris sent me this link:

                      http://doc.pfsense.org/index.php/VoIP_Configuration

                      I think #2 should help me out, but I can't test until next week.

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                      • S
                        Supermule Banned
                        last edited by

                        I know, but I just forward the used ports through PFSense to the PBX, handling the SIP traffic.

                        Good audio and no problems at all.

                        @rugby:

                        @Supermule:

                        I use Askozia PBX in VmWare setup… Works like a charm.....

                        I don't think this has to do with the PBX so much as the natting of SIP ports.  We are going to demo OnSIP in the coming weeks and I saw one of the threads pertaining to SIP nat.

                        I'm just frustrated because this should just work and it's "sort of" working which is worse than not working at all.

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                        • R
                          rugby
                          last edited by

                          I changed the System->Advanced-> Firewall Optimization options to conservative and the phones have stayed registered for an hour which is longer than normal.

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                          • S
                            Supermule Banned
                            last edited by

                            Just change the keep connection alive settings in the SIP phones…..

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                            • R
                              rugby
                              last edited by

                              I could do that, but with 20 phones in 3 states this was much easier to do.

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