[asterisk-users] Dialplan help - MeetMe (or ChannelRedirect) and call monitoring

Edoardo Serra edoardo.serra at webrainstorm.it
Tue Apr 10 04:49:43 MST 2007


Hi guys,
    I need to realize a sort of automatic call monitoring dialplan.

This is exactly what I need:
- A user originate a call
- When the call is bridged (or just before) I need to invite 
automatically a third party to the conversation that should hear the 
audio channel but not speak (it's a monitoring application for a 
callcenter)

The person in charge of monitoring cannot use ChanSpy or whatever
because calls are placed at random hours during the day and its 
telephone should ring when he needs to listen to a call.

I was thining at using a MeetMe in which i'd put both legs of the 
monitored call and the person who should hear the conversation.
Do you have other tips about that ??

Here was my first idea of dialplan to get to it.

[outgoing]
exten => _X.,1,Noop(Call monitored in MeetMe ${CALLERID(num)})
exten => _X.,n,Answer()
exten => _X.,n,Set(_MEETMEROOM=${CALLERID(num)})
exten => _X.,n,Wait(1)
exten => _X.,n,Dial(SIP/trunk/${EXTEN}|G(call-third-party^s1))

[invite-third-party]
exten => s,1,MeetMe(${MEETMEROOM},dAxqa)
exten => s,2,Dial(SIP/extensionformonitoring|G(bridge-all^s1))

[bridge-all]
exten => s,1,MeetMe(${MEETMEROOM},qdx)
exten => s,2,MeetMe(${MEETMEROOM},mqdx)

This setup is not working because I cannot call a Dial again on a 
bridged channel

Here is what I get on Asterisk CLI

  == Starting SIP/180-108c94b0 at call-third-party,s,2 failed so falling 
back to exten 's'
  == Starting SIP/180-108c94b0 at call-third-party,s,2 still failed so 
falling back to context 'default'

Do you have some idea to achieve this kind of result ?
Maybe I can use ChannelRedirect from Asterisk 1.4 ?
Cna you give me a hint on that ?

Tnx in advance

Regards

Edoardo Serra
WeBRainstorm S.r.l.

-- 
Ing. Edoardo Serra
WeBRainstorm S.r.l.
Via Pio Foà 83/C
10126 - Torino

Tel: +39 011 678 100
Fax: +39 011 678 275



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