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    Interesting glitch with asterisk

    Scheduled Pinned Locked Moved Traffic Shaping
    5 Posts 3 Posters 3.4k Views
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    • D
      danswartz
      last edited by

      So I configure the TS with the static IP of my pbx in a flash box, 10.0.0.7.  Make a few test calls.  Puzzled that inbound traffic was going into qVoipDown, but outbound was going into qWanAcks.  Hmmm.  Google around and find a thread involving scott and bill from a couple of years ago that gave me the necessary clue.  To wit: asterisk sets low delay bit by default, which was causing the outbound traffic to go via qWanAcks.  I guess I don't care, since that is also high priority, but I was feeling anal and wanted to understand what was up.  I changed the asterisk config to set the sip and rtp ToS to 0, reloaded asterisk, and voila :)

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      • J
        jstraten
        last edited by

        Could you please share where this can be done in Asterisk? I am having the same problem.

        When I change the setting in sip.conf, Asterisk is showing an error message in the log that it can't set the setting. It seems that Asterisk would have to run as root to do so.

        Thanks,
        Jens

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        • D
          danswartz
          last edited by

          The default location for these is in sip_general_additional.conf, but you tweak them in sip_general_custom.conf.  Like this:

          tos_sip=0
          tos_audio=0

          NB: this is for a freepbx-based system.  If yours is not, you need to figure out where to set these for your setup.

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          • J
            jstraten
            last edited by

            Thanks! :)

            I just tested it and it works great. I am using Elastix which is FreePBX based. Just switched from trixbox.

            I would think that most people who are using Asterisk will want this to make their traffic shaping efforts easier.

            Cheers,
            Jens

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            • K
              kubo999
              last edited by

              @danswartz:

              The default location for these is in sip_general_additional.conf, but you tweak them in sip_general_custom.conf.  Like this:

              tos_sip=0
              tos_audio=0

              NB: this is for a freepbx-based system.  If yours is not, you need to figure out where to set these for your setup.

              Been looking for this solution for a long time. Thanks. Im on the PBXInAFlash asterisk flavor….

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