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    2.0 with IP Phone

    Scheduled Pinned Locked Moved 2.0-RC Snapshot Feedback and Problems - RETIRED
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    • M
      MrHorizontal
      last edited by

      Sounds like the phone is relying on an idle but open state, and once the state disappears from pfSense (it will scrub states after a while) it loses connection and doesn't reset the state until the phone makes a call. I don't think the issue here is with pfSense, but with your trixbox/asterisk configuration - is the phone required to register with the trixbox and also is it using SIP's NAT traversal techniques at all?

      I'm also using trixbox and Linksys SPA942 phones but all the gear is behind pfSense on the LAN side - the only SIP connection through the WAN are the SIP trunks from trixbox to my providers.

      Alternatively, another path to look down is a package for pfSense called siproxd. for some it works better than others - but for me it makes my calls very unreliable and choppy so I don't use it. YMMV.

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      • C
        cmb
        last edited by

        Possibly #1, 2 or a combination of those here.
        http://doc.pfsense.org/index.php/VoIP_Configuration

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        • S
          ScorchedHands
          last edited by

          Thanks for the replies guys!

          I tried to sip proxy package, but it didn't help at all.  I had forgotten to mention that.  I think that disabling the scrubbing should work though.  I do not have any control over the Trixbox config but I assume it is setup to use standard nat-t tricks and such.  I'll have to talk to the trixbox guy about it.  I'm going to try and set the state table optimization for conservative and see how that works.  I'll post back with my results so that other people in the future can find the answer  :)

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          • W
            wallabybob
            last edited by

            If you make an outgoing VOIP call there will be firewall state created so that an incoming VOIP call will go to the same place. After a while the state will timeout and disappear. If there is then an incoming call how will the firewall know where to forward it? Wouldn't you need a Port Forward rule or something like it to cover this situation?

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            • C
              cmb
              last edited by

              @wallabybob:

              If you make an outgoing VOIP call there will be firewall state created so that an incoming VOIP call will go to the same place. After a while the state will timeout and disappear. If there is then an incoming call how will the firewall know where to forward it? Wouldn't you need a Port Forward rule or something like it to cover this situation?

              No, the connection is first initiated outbound (SIP registration and RTP) which lets the server reply back over that open connection (as long as it doesn't get closed, hence use keep alive on your phones, or make sure your state type is conservative).

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              • W
                wallabybob
                last edited by

                @cmb:

                No, the connection is first initiated outbound (SIP registration and RTP) which lets the server reply back over that open connection (as long as it doesn't get closed, hence use keep alive on your phones, or make sure your state type is conservative).

                What is the difference in timeout between Normal and Conservative?

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                • D
                  danswartz
                  last edited by

                  What I would recommend is to use "qualify=yes" for the trunk(s), that should keep the SIP registration active.

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                  • M
                    mloiterman
                    last edited by

                    Try this:

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

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                    • D
                      danswartz
                      last edited by

                      He is already doing static port.

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                      • M
                        mloiterman
                        last edited by

                        Hrmm…yeah, I guess I missed that in the original post.  Well, that solved some VoIP issues for me, so I guess it's worth repeating even if it doesn't work in this specific case.

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                        • chpalmerC
                          chpalmer
                          last edited by

                          Does your Voip provider handle the audio of your calls or do they hand off that task?

                          Who is your provider… (Ill go back and re-read in case I missed...)

                          If your provider hands off the calls then the audio RTP streams will be coming from somewhere else and will be seen by the firewall as an unsolicited attempt to connect.

                          Triggering snowflakes one by one..
                          Intel(R) Core(TM) i5-4590T CPU @ 2.00GHz on an M400 WG box.

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                          • D
                            danswartz
                            last edited by

                            Not necessarily - the moment the IP phone sends any outbound RTP, this should establish a state table entry.  Also, he was complaining about receiving calls, which is a SIP issue, not RTP.

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                            • chpalmerC
                              chpalmer
                              last edited by

                              My "will be seen" should read "could be seen" regarding the firewall…  But I see your point...

                              Just throwing an idea out there.  Initially got me on a PBX I tried to run internally...

                              But after re-reading...  I had to turn the keep alive on with the ata's at the office behind the 1.2.3 install I have there to allow my Freeswitch install I have here to reach it after the states wanted to expire.

                              My providers send some form of keep alive from their servers to my home ata's that I can not duplicate on Freeswitch here. However it shouldn't matter what side it comes from.

                              You should not need port forwarding on the ata/ip phone side to make this work.

                              Triggering snowflakes one by one..
                              Intel(R) Core(TM) i5-4590T CPU @ 2.00GHz on an M400 WG box.

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                              • M
                                mgaudette
                                last edited by

                                Try to configure your phone to re-register every 60 seconds. That will fix it. Sometimes it's a "re-register" field, sometimes it's called "Register expiry" or something like that.

                                As long as it re-registers faster than the firewall forgets your state, you'll be be ok.  60 seconds seems to do the trick on pfSense vanilla.

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                                • S
                                  ScorchedHands
                                  last edited by

                                  Well I got the problem sorted out for the most part, but now I have a new problem.  The tftp-proxy is missing on 2.0 BETA1.  I'm going back through the snapshots to see when it went missing, but I haven't found it yet.  Does anyone have a copy of "/usr/local/sbin/tftp-proxy" that I can use?  I'm getting

                                  inetd[30822]: cannot execute /usr/local/sbin/tftp-proxy: No such file or directory

                                  It's in the System Log whenever I try to push tftp traffic, so it appears that it's configured properly, just not there lol

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                                  • S
                                    ScorchedHands
                                    last edited by

                                    Oh, and I'm also trying to get it to use tftp over an ipsec vpn. It doesn't seem to want to route the tftp traffic.  I've confirmed that tftp is working on the remote network.

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                                    • S
                                      ScorchedHands
                                      last edited by

                                      I got tftp to work over the ipsec, I had to disable the tftp proxy.  well I couldn't disable it actually, i just switched it to OPT1 to it wouldn't interfere with LAN traffic.

                                      so that's two things broken, missing tftp-proxy and the gui on the advanced page is broken.  it doesn't highlight the enabled interfaces so it always looks completely disabled.

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                                      • C
                                        cmb
                                        last edited by

                                        I fixed the select issue, there's a ticket open on the missing binary. You don't want or need the TFTP proxy in most routing scenarios including IPsec, only with NAT is that needed generally.

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                                        • S
                                          ScorchedHands
                                          last edited by

                                          nice, yea I'm aware it isn't needed for ipsec.  but i want to say that it being on and set to LAN made tftp not work over my ipsec tunnel so it must be doing something there.  or maybe i'm just misunderstanding how to configure it.  for an ip phone on LAN to work with a tftp server on the internet/WAN (bad practice i know), does the proxy have to be set to LAN or WAN?

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                                          • C
                                            cmb
                                            last edited by

                                            Proxy is set on the interface(s) where TFTP traffic is initiated.

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