[asterisk-users] Dialplan help - MeetMe and call monitoring
Edoardo Serra
edoardo.serra at webrainstorm.it
Tue Apr 10 04:25:06 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/${EXTEN}|G(call-third-party^s^1))
[invite-third-party]
exten => s,1,MeetMe(${MEETMEROOM},dAxqa)
exten => s,2,Dial(SIP/extensionformonitoring|G(bridge-all^s^1))
[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 ?
Tnx in advance
Regards
Edoardo Serra
WeBRainstorm S.r.l.
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