[asterisk-users] sip to sip unplanned conference! help!!

[RoLaNd RoLaNd]

first of all my topology is as such:Softphones<<-->> asterisk <<--> sipurasoftphone with peer number 100, calls another softphone with peer number as 200. (both has asterisk as gateway)relevant extensions.conf:

exten => _1XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so it will ring 3 times
exten => _1XX,2,VoiceMail(${EXTEN}@default) ; direct 2 voicemail box if line is busy or unavailable
exten => _1XX,3,HangUp()
exten => _2XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so it will ring 3 times
exten => _2XX,2,VoiceMail(${EXTEN}@default) ; directs to voicemail box if line is busy or unavailable
exten => _2XX,3,HangUp()


relevant sip.conf:
[200]
type=friend
host=dynamic
secret=1234
context=spa
mailbox=102 at default

[200]
type=friend
host=dynamic
secret=1234
context=spa
mailbox=102 at default


in the meantime, an incoming call comes through Sipura which is directed to:

[incoming-samer]
exten => 201,1,Answer() ; Answer inbound calls
exten => 201,2,Playback(silence/1)
exten => 201,3,Background(joyce) ; input an extension
exten => 201,4,WaitExten(8)
exten => 201,5,Dial(SIP/220,15)
exten => 201,4,Wait(8)
include => spa
exten => 201,n,Hangup()
suddenly, the first conversation between 100 and 200, hears the attendant audio message "joyce" welcoming the caller(the one calling sipura in a completely different call) and listens to the entire conversation that the incoming caller is having..

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