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    Hosted VOIP and pfSense

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

      @danswartz

      It is because port 5060 (and I think another one of two) are not covered by the Automatic rule. I read that in the pfSense Definitive Guide book but I think it's also in the docs somewhere.

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      • D
        danswartz
        last edited by

        I think you misread that.  What is treated specially for port 5060 is pfsense not doing the rewriting of it.

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        • R
          rugby
          last edited by

          I'm seeing the same thing with our hosted PIAF setup.  We have 4 SPA-942 phones and 1 Aastra 57i CT and they randomly unregister over the course of the day.  Siproxd didn't do anything, manual NAT works until the phones try and re-register and then they fail.

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          • M
            mst
            last edited by

            Did you put

            nat=yes
            externip=xxx.xxx.xxx.xxx
            externhost = mypbx.mydomain.com
            localnet=192.168.1.0/255.255.255.0
            externrefresh=10

            in SIP_NAT.conf

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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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