[asterisk-dev] Re: asterisk-dev Digest, Vol 21, Issue 72

jemmy_12345 frank jemmy_12345 at yahoo.com
Sun Apr 23 11:05:01 MST 2006


Hi All
   
  I want to implement a function as.
  user  ---> call ( from telco) --> asterisk ---> IVR -- SIP 1.
  after that, SIP1 transfer to SIP2 (unattendant or attendant transfer). i want to SIP1 hear stream sound data of call conversation between user and SIP 2 (don't used call conference)
   
  SIP3 want to hear stream sound data of user and SIP2 conversation, can be press DTMF keys as: form example: *8401 ( 401 as username of SIP2).
   
  could you like to help me to implement that function.
   
  Best regards

		
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