MixMonitor and ChanSpy: Whisperer's voice not recorded

Hi, we are using ChanSpy for whispering and also using MixMonitor for recording the conversation. Problem is that the whisperer’s voice is not recorded in the audio file, only the agent and the caller, not the supervisor. Any ideas why?

We first do MixMonitor, then if the supervisor needs, he may whisper (using ChanSpy.)
So it is, usually, first MixMonitor -> then ChanSpy.

Here is the CLI log from the console:

== Using SIP RTP CoS mark 5 -- Executing [210@call_manager:1] Answer("SIP/212-00000073", "") in new stack -- Executing [210@call_manager:2] Set("SIP/212-00000073", "_CallContextId=1377250866.115") in new stack -- Executing [210@call_manager:3] Set("SIP/212-00000073", "_CallStartTime=2013-08-23 14:41:06") in new stack -- Executing [210@call_manager:4] Set("SIP/212-00000073", "_LastMenu=None") in new stack -- Executing [210@call_manager:5] Set("SIP/212-00000073", "_Parameters=NEW-PSTN-None") in new stack -- Executing [210@call_manager:6] Set("SIP/212-00000073", "_CallType=Inbound") in new stack -- Executing [210@call_manager:7] NoOp("SIP/212-00000073", "") in new stack -- Executing [210@call_manager:8] AGI("SIP/212-00000073", "agi://192.168.3.8:4574/callmanager,1377250866.115,2013-08-23 14:41:06,NEW-PSTN-None,None,Inbound") in new stack -- AGI Script Executing Application: (AGI) Options: (agi://192.168.3.8/mblrendevous, 1377250866.115, None) -- Playing '/var/lib/asterisk/sounds/mblrendevous/Welcome' (escape_digits=0123456789*#) (sample_offset 0) -- Playing '/var/lib/asterisk/sounds/mblrendevous/E_MainMenu' (escape_digits=0123456789*#) (sample_offset 0) -- Playing '/var/lib/asterisk/sounds/mblrendevous/E_TransferToAgent' (escape_digits=) (sample_offset 0) -- <SIP/212-00000073>AGI Script agi://192.168.3.8/mblrendevous completed, returning 0 -- AGI Script Executing Application: (System) Options: (mkdir /var/spool/asterisk/monitor/Inbound/23082013/) -- AGI Script Executing Application: (MixMonitor) Options: (/var/spool/asterisk/monitor/Inbound/23082013/1377250866_115~1377268706.wav, b) == Begin MixMonitor Recording SIP/212-00000073 -- AGI Script Executing Application: (Queue) Options: (MeezanRendevous) -- Started music on hold, class 'mblunison', on SIP/212-00000073 == Using SIP RTP CoS mark 5 -- Called SIP/1003 -- SIP/1003-00000074 is ringing -- SIP/1003-00000074 answered SIP/212-00000073 -- Stopped music on hold on SIP/212-00000073 == Using SIP RTP CoS mark 5 -- Executing [3331003@call_manager:1] ChanSpy("SIP/1002-00000075", "SIP/1003, qB") in new stack == Spying on channel SIP/1003-00000074 [Aug 23 14:41:34] NOTICE[25896]: app_chanspy.c:415 start_spying: Attaching SIP/1002-00000075 to SIP/1003-00000074 [Aug 23 14:41:34] NOTICE[25896]: app_chanspy.c:415 start_spying: Attaching SIP/1002-00000075 to SIP/1003-00000074 [Aug 23 14:41:34] NOTICE[25896]: app_chanspy.c:415 start_spying: Attaching SIP/1002-00000075 to SIP/212-00000073 == Done Spying on channel SIP/1003-00000074 == Spawn extension (call_manager, 3331003, 1) exited non-zero on 'SIP/1002-00000075'

Any help would be really appreciated.

p.s. we are using Asterisk version 1.6.2.22, if that helps.

By any chances have you find a way to do that properly please ?
If you can share your knowledge i’m sure it can be useful to a lot of people.

Hi,

According to your shared logs you are recording(MixMonitor) on customer call leg and whispering on agent’s call leg. According to the Whispering method supervisor voice will only audible to agent and that voice(RTP packets) won’t transmit to customer call leg. That’s why in call recordings customer and agent voice is available.