Extension behind NAT

Discussion in 'General FusionPBX Help' started by Jeroen Hermans, Oct 12, 2017 at 4:15 PM.

  1. Jeroen Hermans

    Jeroen Hermans New Member

    Hi all,

    I know i am missing something trivial here. I have read a lot about NAT and i do not seem to get this right.
    I have a system running:
    phone--->NAT router--->internet--->fusionPBX (without NAT)--->trunk provider (no NAT)

    Now, when i make a call with my phone, i see in the following SIP packets (heavily redacted):
    INVITE sip:0031xxxxxx@tenant.voip.domain.nl:5080 SIP/2.0
    Via: SIP/2.0/UDP;branch=z9hG4bK1173805850;rport
    From: <sip:202@tenant.voip.domain.nl:5080>;tag=712368681
    To: <sip:0031xxxxxxxxxxx@tenant.voip.domain.nl:5080>
    Contact: <sip:202@>
    User-Agent: Grandstream GXP2130

    Obviously the Via and Contact contain the NAT address of my phone (and not the public address of the NAT router). The result is that the RTP stream is send to the address which, of course, ends up nowhere...

    The weird thing is: SOMEtimes the calls DO get through and the public ip of the NAT router of the extension is used.

    What am i doing wrong here? Please mind: FusionPBX is NOT behind nat... the (remote) phone is.
    Thank you very much in advance.

  2. Adrian Fretwell

    Adrian Fretwell New Member

    Don't have any knowledge of Grandstream, but if those phone are similar in configuration to Yealink there should be some settings around NAT Traversal. I tend to use a STUN server, there are a few publicly available ones on the internet. With STUN enabed the Yealink phones correctly set the contact headers and the In IP4 in the SDP body.

    In the Internal SIP profile there are also some NAT settings to allow FreeSwitch to "Fix" some NAT problems, I believe they are normally defaulted true.
  3. EasyBB

    EasyBB Member

    This is the most common scenario.....

    Did you check SIP ALG setting in the router?

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