Hi, I am trying to integrate OpenSIPS with Asterisk for playing media and IVR treatment. Below is the copy of the sip.conf file and extensions.conf. When i dial 12345 from my softphone (Bria), the call drops without playing the media file in asterisk which is the expected outcome. I am seeing 401 unauthorized message in the asterisk console when the invite is sent. Any help would be appreciated ? Thank you
sip.conf
[general]
context=from-sip
bindaddr=0.0.0.0
bindport=5060
allowguest=no
[opensips]
type=peer
host=192.168.1.201
context=from-sip
insecure=invite
allow=ulaw;alaw;gsm
deny=0.0.0.0/0.0.0.0
permit=192.168.1.201/255.255.255.255
extensions.conf
[from-sip]
exten => 12345,1,Answer
exten => 12345,n,NoOp(OpenSIPS-Integration)
exten => 12345,n,playback(demo-thanks)
exten => 12345,n,Hangup()