7 thoughts on - Can Anyone Help With A Quick App_record.c Module Improvement And Can Explain Over-riding Modules?

• Just a quick and dirty thought, try the MONITOR application.

  Pseudo-code:

  Anchor-point PLAYBACK (“press or say”)
  MONITOR (use the split audio files mode, not the mixed – this way you can roughly separate which side did the “talking”)
  READ (audio file “1 to 5”, try to grab one digit)
  STOPMONITOR
  IF (READ variable timed-out, send the incoming half of the monitor file to Google Speech)
  Playback (some sound effect to indicate “thinking” on the Asterisk side – user feedback is good)
  Check Google Speech result against a white-list
  IF filtered result was not a valid option
  PLAYBACK “I didn’t understand that”
  GOTO to Anchor-point
  ELSE
  Goto next step using valid decoded speech data ELSE
  Check DTMF result against a white-list
  IF filtered DTMFresult was not a valid option
  PLAYBACK “I didn’t understand that”
  GOTO to Anchor-point
  ELSE
  Goto next step using valid decoded DTMF data Catch-all, should never get here.

  /Pseudo-code

  Don’t forget to filter your user sourced data against your white-list, always assume users are hostile, this is part of the total picture of defence-in-depth.

  -Tim

• Oh, what a good idea! That’s exactly the kind of lateral thinking I
  was hoping someone would come up with.

  I thought it was called MixMonitor, and tried to wrap my head around it but couldn’t.

  I’ll give this a go tomorrow and let you know what I come up with!

  Many thanks,

  Jonathan

  T

• MixMonitor is related, but different (and as the name suggests, automatically mixes the two channels, so I think Tim’s suggestion to use Monitor is much better.

  Note that you may well need to use the ‘b’ option with Monitor, to make sure you can record when there’s no bridge between two channels.

  Please do report back – this is a useful feature.

  Antony.

• Also, be aware that by creating an audio file, you may need to insert a pause in your code before the file is:

  1) written
  2) flushed from cache to disk
  3) registered as available to be opened by the OS

  I have seen this take over 2 seconds before on a sluggish machine. You can speed this up a bit by putting the recordings in a RAMDISK
  partition on the host – but be careful that you only use short recordings and clean them up after they are not needed any more.

  If that’s still not fast enough, there’s the Google Speech streaming API, but I’m not up to snuff on that – essentially you’d need the functionality of the monitor split to only stream the remote user’s voice, then you’d need to pipe that to a Google Speech API tunnel. That’s probably not something you can hack away at with simple Asterisk dialplan applications.

  Cheers,

  -Tim

  —

• Thanks – my host uses SSD and everything seems pretty quick, but I’ll give it a 1 second pause.

  Funnily enough, I had just found an old reply from last year to another similar question:

  So I had a look and found this:

  https://stackoverflow.com/questions/34026698/asterisk-write-plugin-to-catch-voice-stream

  And read this:

  https://wiki.asterisk.org/wiki/display/AST/Asterisk+15+Application_EAGI

  There’s a few knowledge gaps, but I think with a few days reading and the great help here, we might have a solution 🙂

  This is all very helpful – if anyone else feels like wading in, please do.

  Many thanks!

  —

• Hello.

  A little sub from my dialplan:

  [sub-Read]
  exten => s,1,NoOp(Read)
   same => n,Set(LOCAL(tmp_record_file)=/tmp/asterisk-in/${EPOCH})
   same => n,Monitor(wav16,${tmp_record_file},o)
   same => n,Read(tmp_ext,${ARG2},${ARG3},${ARG4},${ARG5},${ARG6})
   same => n,StopMonitor()
   same => n,NoOp(ReadStatus=${READSTATUS})
   same => n,Gotoif($[ ${LEN(${tmp_ext})} > 0 ]?end)
   same =>
  n,AGI(agi-ruvoip-net.php,speeddial-voice,${ARG1},${tmp_record_file}-in.wav16)
   same => n,NoOp(Voice recognition result: “${agi_result}”)
   same => n,Gotoif($[ “${agi_result}” != “found” ]?end)
   same => n,Return(${agi_call_exten})
   same => n(end),return(${tmp_ext})

  21.01.2018 2:57, Jonathan H пишет:

  —

• Hi Dmitry and Tim (and everyone else with input into this thread)

  Just wanted to thank you all; with your guidance, I’ve managed to bolt something very clean and efficient together using Dmity and Tim’s templates, piped into the ding-dong npm node package, which calls Google Speech API node package.

  What I particularly like about DingDong is that it’s well documented insofar as it’s so simple, it barely needs documentation!