Whisper seems to be broken. Barge works ok. Listen works ok

Asterisk 1.8.3 on a x86_64 running Linux with FreePBX 2.9.0rc1.0 plus FOP2

When I use chanspy to Listen to a call it works as I expect: No interruption in the established call and Eve can hear both parties.

Likewise when I use Barge to listen to a call. The established call continues, Eve can hear both parties and talk to both parties.

However, Whisper replaces the targeted extension’s receive audio path with the audio channel from Eve. Eve can hear both parties and can talk to the target party. However, the targeted party can no longer hear the other party. When Eve hangs up (or switches to another channel, or is listening to the announcement about the channel), the targeted party can once again hear the other party.

If I use * to toggle through other channels, subsequent channels don’t mute the audio on the established channel.
I am new to this, however, I can’t believe this is intended behaviour. Any suggestions please?

Cheers now
Dave

  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
       > Channel SIP/5102-0000016a was answered.
       > Launching ChanSpy(SIP/5103,w) on SIP/5102-0000016a
    -- <SIP/5102-0000016a> Playing 'beep.ulaw' (language 'en')
    -- <SIP/5102-0000016a> Playing 'spy-sip.ulaw' (language 'en')
    -- <SIP/5102-0000016a> Playing 'digits/5.ulaw' (language 'en')
    -- <SIP/5102-0000016a> Playing 'digits/1.ulaw' (language 'en')
    -- <SIP/5102-0000016a> Playing 'digits/0.ulaw' (language 'en')
    -- <SIP/5102-0000016a> Playing 'digits/3.ulaw' (language 'en')
  == Spying on channel SIP/5103-00000166
  == Done Spying on channel SIP/5103-00000166

Howdy,

That’s not intended. So that I undertand, only the first channel whispered to has the incorrect audio path, yes? Subsequent channels, accessed via *, are correct?

Hi Malcolm,

Yes. Certainly the audio on the first channel is wrong.
I am not completely certain of the reproducible conditions in which the audio channel becomes correct again. I could do more testing…

Howdy,

That’s not desired behavior, so the issue tracker should come into play:

asterisk.org/developers/bug-guidelines

Certainly, the more you can nail down the issue’s specifics, the more it helps.

Cheers.

I have been messing with this for a day or more now and at the moment can’t reproduce the scenario in which I cured the issue by * ing through channels. Nevertheless, I believe the issue remains which is the key point. I am about to finish work for the week, so haven’t much time today.

I have read the pages you linked to. I understand the basis of how these things work, but didn’t see an area to file a bug report. Could you indulge me with another pointer please?

Cheers

UPDATE I see it. issues.asterisk.org Wow colourful! :smile: