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  1. #1
    Untangler
    Join Date
    May 2008
    Posts
    568

    Default Needed wiki for VOIP

    Hi

    I am having trouble getting voip running through Untangle. There are lots of posts to read and I think I will get it to work but it would be good if there was a wiki article that had this info in one place. It would be great if someone had time to do this. If not a full wiki maybe at least a sticky item to find it easily. Suggestions to include asterisk setting and some other popular services.

    I think this would be really helpful to a lot of people, not just me.

    Thanks,
    Don

  2. #2
    Master Untangler
    Join Date
    May 2008
    Location
    Ohio
    Posts
    161

    Default

    Whose voip offering are you using ? Are you running UT a a bridge or router and do you have QOS turned on the UT box ?

  3. #3
    Untangler
    Join Date
    May 2008
    Posts
    568

    Default

    Hi I am using Elastix and Untangle 6 in router. I had it working before with 5.2 I think but am revisiting this. I think it would be great if there was one place to get at least the basic info. Looks like UT is now acting as some sort of sip gateway from reading some posts. I guess it is just a matter of setting asterisk to not renat stuff. Going skiing for a couple hours and look at it latter.

    Thanks,
    Don

  4. #4
    Untangler
    Join Date
    May 2008
    Posts
    568

    Default

    What type of natting is Untangle doing to sip & rtp . Is it a total application gateway? Or is it some type of cone nat, I cant seem to figure it out. Maybe it is not an Untangle problem but I still need to know how Untangle is handling these things. I think it has to do with nat because I can get audio both ways inbound but one way outbound.

    Thanks,
    Don

    I am upgraded to 6.0.1
    Last edited by donhwyo; 12-15-2008 at 11:33 AM.

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