[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:[EMAIL PROTECTED]

[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/[EMAIL PROTECTED]
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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