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  1. FreeSWITCH
  2. FS-9939

Audio cut off at the begin of the verto call to sip external voicemail

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    Details

    • Type: Bug
    • Status: Closed
    • Priority: Minor
    • Resolution: No Reporter Response
    • Affects Version/s: 1.6.13, 1.8
    • Fix Version/s: None
    • Component/s: mod_sofia, mod_verto
    • Labels:
      None
    • CPU Architecture:
      x86-64
    • Kernel:
      Linux
    • Userland:
      GNU/Linux
    • Distribution:
      Debian
    • Distribution Version:
      Debian 8 jessie
    • Compiler:
      gcc
    • FreeSWITCH GIT Revision:
      232ed6e5d2
    • GIT Master Revision hash::
      232ed6e5d2

      Description

      I have audio cut off at the begin of the verto call to FreeSwitch that redirect to sip external voicemail (Access voicemail mailbox) .

      This happen when I use PCMU at verto codecs and sip codecs (if i use opus at verto codecs, there is no issue, but this causes audio transcoding) .

      At dialplan i used the example "Bridging from WebRTC (mod_verto) to PSTN/ITSPs" from https://freeswitch.org/confluence/display/FREESWITCH/mod_verto.
      I notice if i remove the playback action, there is no issue. But I need the playback action to send rtp packets to verto client.

      I simulate this using another FreeSwitch as external voicemail server and I only listen "id followed by pound" from the initial message of voicemail ("Please enter your id followed by pound").
      The log of this call is at https://pastebin.freeswitch.org/view/507fa115

      I tried to use an audio file (sounds/en/us/callie/ivr/8000/ivr-say_name.wav with ~2 seconds) instead of silence_stream.
      When i make the call from verto client, i ear the audio file, then no audio for ~2/3 seconds and then i ear "id followed by pound" (audio cut off from voicemail initial message "Please enter your id followed by pound").
      The log of this call is at https://pastebin.freeswitch.org/view/e130e172 .

      I checked if i have the variable answer_delay at vars.xml and i don't have it.

      I tried another test, I add a sleep of 2000 at voicemail server (after the answer application) and there is no audio cut off.
      But there is silence of 3/4 seconds between the ivr-say_name and the initial message from voicemail without audio cut off.

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            • Assignee:
              mikej Mike Jerris
              Reporter:
              jose.lopes Jose Lopes
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              Dates

              • Created:
                Updated:
                Resolved: