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  1. FreePBX
  2. FREEPBX-20232

AMI Generated Calls to outbound SIP channels (External to NAT) intermittently do not send outbound audio

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    • Bug Tracker:
      Customer Issue
    • ToDo:
    • Asterisk Version:
      13.17.0 / 16
    • Distro Version:
      FreePBX 14

      Description

      AMI generated calls from local PHP script intermittently fail to send outbound audio to external party while external party is able to send audio to local SIP party.  If the AMI generated call stays internal ie: SIP/100 to SIP/200 calls has bidirectional audio everytime.  Calls placed directly from the SIP device to internal / external (upstream SIP provider over internet work everytime.  Only AMI generated calls, approx 90% of the time, will not send audio from SIP/100 to internet upstream provider to external device.

      In all cases RTP traffic looks correct to me.  

       

      These are my pastebin logs for a call that worked (2 way audio) and one that the outbound audio failed, these were placed a few minutes apart on the same server without any setting changes.

      Click to Call WORKED: https://pastebin.freepbx.org/view/babc9da1 1

      Click to Call FAILED: https://pastebin.freepbx.org/view/334d0c8e

      I had created a topic on community.freepbx.org without much success however my logs and RTP traffic are posted there.  A more complete description of the issue is posted on the community forum.

      https://community.freepbx.org/t/ami-generated-calls-intermittent-one-way-audio-normal-calls-placed-from-phone-work-all-the-time/59972

      It seems to me this must be a bridging issue.

       

      call.php.txt is obviously just call.php on the server and it was is used for running the AMI process

       

       

       

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              • Assignee:
                Unassigned
                Reporter:
                mfranz83 mf83
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                  Updated:
                  Resolved:

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