asterisk users - Nov 2006 - Asterisk as a SIP client, Need to auto-answer

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@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@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


 
---------------------------------
Sponsored Link

$420,000 Mortgage for $1,399/month -   Think You Pay Too Much For Your Mortgage?
Find Out!
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20061115/8f924a34/attachment.htm