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    SIP hangs after a while

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    • H
      hoba
      last edited by

      What is "a while"?  You might try to add a long state timeout for these firewallrules (edit the rules, statetimout option is hiding behind one of the advanced option buttons). Maybe the state times out when the connection is idle for some time.

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      • B
        Boardercrime
        last edited by

        How long i should set the timeout ?
        I think after 1-2 minutes it hangs..not sure.

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        • H
          hoba
          last edited by

          1-2 minutes? that sounds like something is closing the state. that's just too short to be a problem with state timout imo but try something like an hour just to test if something changes.

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          • B
            Boardercrime
            last edited by

            I seems the timeout settings is not the problem.
            Any ideas ?

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            • H
              Helix26404
              last edited by

              In Asterisk, try setting qualify=yes for the extension that is affected. This will force Asterisk to send "keep alive" packets to the device.

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              • B
                Boardercrime
                last edited by

                I've got this setting already.

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                • S
                  SpLord
                  last edited by

                  any solution yet, same problem here.

                  After resetting the State table everything is working for round a day, then you has to reset the states

                  Carsten

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                  • H
                    hoba
                    last edited by

                    You probably need some kind of keepalive setting for your SIP-client. I have seen devices that have a setting for this (like the AVM Fritzbox for example). Check the manuals of your device to see if it supports such a setting.

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                    • S
                      SpLord
                      last edited by

                      @hoba:

                      You probably need some kind of keepalive setting for your SIP-client. I have seen devices that have a setting for this (like the AVM Fritzbox for example). Check the manuals of your device to see if it supports such a setting.

                      keepalive is active, i'm using an asterisk server which connects too some sip providers.

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                      • S
                        Slam
                        last edited by

                        More settings for you guys to play with if you have a Sipura/Linksys ATA

                        (under NAT Support Parameters):
                        STUN Enable=yes, STUN Test Enable=yes, Substitute VIA Addr=yes, Send Resp To Src Port=yes, NAT Keep Alive Intvl=15, other VIA parameters=no

                        (under Line 1):
                        NAT Mapping Enable=yes, NAT Keep Alive Enable=yes

                        Regards

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                        • H
                          hoba
                          last edited by

                          Try this solution: http://forum.pfsense.org/index.php/topic,4364.msg26872.html#msg26872

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                          • S
                            SpLord
                            last edited by

                            Nope doesn't work.

                            Is there a possibility to trigger a state table reset when a re-dial(PPPoe) occurs?

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                            • M
                              maldex
                              last edited by

                              Try to decrease the register times to 60seconds. PFsense, along with some expesive-firewalls, have UDP timeouts of 30/60 seconds… after 60 seconds the incomming INVITE will be dropped.

                              Using STUN doesnt solve the problem. stun is only used to let the phone know the public(masqueraded) address, and how it can open up UDP sessions.

                              the public IP is needed because SIP (which is osi-layer7) does also contain the IP adress, and some SIP-devices will answer only on that and not on the layer3 ip...(workaround in asterisk is NAT=Yes)

                              another good idea is to create a NAT rule which does static-port-mapping on the SIP & RTP sessions so that port 5060 stay's always 5060.....

                              things i haven't checked yet for myself:

                              • SIP over TCP. TCP-sessions have much longer timeouts...but is rarely supported
                              • Conservative mode.

                              good luck

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