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