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  1. FreePBX
  2. FREEPBX-9158

RTP port range changes after updating SIP Settings module

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    Details

    • Type: Bug
    • Status: Closed
    • Priority: Minor
    • Resolution: Fixed
    • Affects Version/s: 12
    • Fix Version/s: 12
    • Component/s: Asterisk SIP Settings
    • Labels:
    • Environment:

      FreePBX distro on VMWare, 6.12.65-25.

    • ToDo:
    • Asterisk Version:
      6.12.65-25
    • Distro Version:
      6.12.65-25
    • Distro:
      FreePBX Distro

      Description

      We use a non-standard port range for RTP for our distro, and I noticed in at least 2 instances that I've updated the SIP Settings module and the port range has been changed back to 10000-20000. This causes calls to be placed as normal but no audio comes through (naturally).

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              • Assignee:
                xrobau Rob Thomas
                Reporter:
                cramermp Michael Cramer
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                Dates

                • Created:
                  Updated:
                  Resolved:

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