3cx Invite "VIA" SIP header address is 192.168.1.90

Discussion in '3CX Phone System - General' started by gp3cx, Jun 1, 2017.

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  1. gp3cx

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    running 15.0.0 with dynamic internet address. (watching with wireshark).....remote sip extentions when called, get sip invites with non routable VIA header address of 192.168.1.90. of course the call fails. this cant be right.....? please help
     
  2. lneblett

    lneblett Well-Known Member

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    where is the capture being done and what does the Connect indicate?
     
  3. gp3cx

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    capturing at remote extension end. im a bit of a novice..could you expand on "what does the Connect indicate" thanks for responding!
     
  4. GiannosC_3CX

    GiannosC_3CX Guest

    Hi gp3cx,

    I have sent you a p.m. please check your inbox.
     
  5. lneblett

    lneblett Well-Known Member

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    IN the SIP messaging between the devices, in addition to the VIAs, there is a section in the Message header called CONTACT (sorry, I labeled wrong earler). The CONTACT info tells the device that received the message as to what IP it should send its response to. Addtionally, there is a similar header in the Message Body where the SDP information is located which tells the receiving device where to send its audio stream and to what port. The phone will also send 3CX a similar message with CONTACT information for 3CX to use.
     
  6. gp3cx

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    isnt the contact info the "sip" address of the invite origin? the VIA address is the one actually used for the response to go to .....my remote extension is a spectralink 8440..... i can see it trying to respond to the VIA address...192.168.1.90
     
    #6 gp3cx, Jun 2, 2017
    Last edited: Jun 2, 2017
  7. lneblett

    lneblett Well-Known Member

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    The VIA is used to show the paths that the messaging took along with certain attributes so that the devices in the path know how to communicate with one another as well as keep track of calls - branch (as many could be taking place). So, for instance, if a call originated from your provider the VIA might have their IP as would the Contact Header. When it hits 3CX and gets relayed, then a new VIA is added showing 3CX and the aforementioned attributes which are then relayed to the phone. The VIAs continue to build as the paths are added to get to the desired destination....but each path inserts a CONTACT.

    So, in a way, yes, you are correct in that ultimately the VIA IP will be where the messaging will be received, but there are some obstacles that stand in the way.

    As you noted, you see a non-routable IP so to get around this, the CONTACT header can be manipulated such that it will show the IP (public) so that the forwarding can be accommodated. This same functionality is also in the SDP headers so that audio streams can find their way to the correct media server and port and the media server may not be the same as the SIP server, so it may very well have a different public IP in the SDP.

    So, look at the CONTACT header in the INVITES,


    .
     
  8. gp3cx

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    Thanks for the great info. my 8440 phone will only respond to the VIA address and I have no option to modify it. So my basic problem is getting 3CX to put the right address in the VIA header. seems there should be a parameter to make 3cx do this right.
     
  9. lneblett

    lneblett Well-Known Member

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    I assume the 8440 is a Spectralink?

    When setting up in 3CX, set the interface in provisioning to STUN. In the 8440, set the NAT.IP to the public IP of 3CX. Set nat.keepalive.interval to 30, nat.signalport to 5060 and the nat.mediaportstart to whatever port desired.

    You may need to set the remote router up for port forwarding of the above and you should refer to the admin manual to determine how best to secure the 8440 from accepting SIP signaling from all but the 3CX system. The guys out their scanning for open SIP ports will drive you crazy otherwise. You could change the signalling port, but they will ultimately find it.
     
  10. gp3cx

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    thanks! tried that..... 8440 still tries to reply to sip VIA address 192.168.1.90
     
  11. lneblett

    lneblett Well-Known Member

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    Ensure router at both ends have SIP ALG off and then do packet capture and post.
     
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