SIP communication with outside server ok with PF 1.2.3 but not ok with PF 2.0.1



  • Hello,

    I replaced a PF 1.2.3 box with a PF 2.0.1. Everything is ok (NAT, rules, proxy, snort, dhcp, routes, …) except one thing : the SIP (VOIP) communication between the server inside my LAN and the external SIP server doesn't work anymore.

    It's like my server was saying to the SIP server : send the reply to my internal IP.

    I kept all default parameters on 2.0.1 (except for NAT, rules, proxy, snort, dhcp that are configured the exact same way as for the 1.2.3 box).

    I tryed to search on internet and on PF forum but couldn't find any help.

    Do you know if there is a default parameter on 2.0.1 that could block my SIP communication ?

    Thank you for your help.

    Jacques



  • See #1 here.
    http://doc.pfsense.org/index.php/VoIP_Configuration

    You're probably one of the very few that need manual outbound NAT with static port in your case. That's the only difference between the versions with SIP.



  • I'll have a look at it and let you know.

    Thank you for your help.



  • Quite often one can also solve this issue by upgrading the VoIP firmware and/or software.

    I believe it's only in those rare cases where both VoIP end-devices are behind symmetric NAT or one side is using antiquated software, where one would have to resort to using manual NAT and configure 'static port'



  • @cmb:

    See #1 here.
    http://doc.pfsense.org/index.php/VoIP_Configuration

    You're probably one of the very few that need manual outbound NAT with static port in your case. That's the only difference between the versions with SIP.

    This is the key.  I've got a VOIP provider that we use pfsense with and we found that we had to enable manual outbound NAT to get more than a single device to register to their servers.  Works great after you do this.  We also found that if you disable the STUN feature on the handset that the issue would go away.

    The only lingering issue we are having is that we have page adapters (basically a sip device that only gets used when the floor is paged) that will disconnect from the server after a while and the only for sure way to get it working again is to reboot the device. Once we get this resolved we will be golden.



  • I had to use AON too for SIP sucessful registering.

    Best

    Kostas


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