[asterisk-users] Asterisk as a SIP client, Need to auto-answer

[Ehsan Khosrowshahi]

Hi all,
   
  I want to initiate a call from the asterisk to an extension, where I will forward 
  the asterisk side to another extension later (to the conference extension). I can 
  initiate a call uning originate call from an extension to the desired extension, 
  but it would need someone from the originator extension to answer the phone. How 
  can i register an extension to asterisk where it automatically answers the phone 
  and creates a channel where I may be able to redirect that channel later to the 
  conference room.
   
  This is what I have done and didnt work:
   
  SIP.conf
register => 70000:70000 at 191.21.21.21
  [70000]
type=friend
auth=md5
username=70000
secret=70000
callerid=70000
host=191.21.21.21
reinvite=no
canreinvite=no
qualify=1500
nat=yes
   
  and in Extension.conf I got:
  exten => 70000,1,Answer
  
and when I originate a call using Manager API with these parameters:
  Channel: SIP/70000 at 70000
CallerID: 70000
Exten: Any number
  I got the following error in asterisk CLI:
    == Manager 'manager' logged on from 191.21.21.21
    -- Got SIP response 482 "Loop Detected" back from 191.21.21.21
       > Channel SIP/0041435215309-3c5a was never answered.
  == Manager 'manager' logged off from 191.21.21.21
  
I want to create a dump connection between a dump extension to any extension then 
  redirect the channel from the dump extension side to the conferece. but How can i make the dump  extension to auto-answer and create a channel when I Originate Call using manager API?
   
  Best
Ehsan


 
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