My setup uses a sip trunk to deal to Skype gateway that resides on the same machine.
This sip trunk is configured this way :
disallow=all
allow=ulaw&alaw&ilbc&speex
canreinvite=no
context=from-trunk
dtmfmode=rfc2833
host=127.0.0.1
incominglimit=1
nat=never
port=5070
qualify=yes
secret=xxxxxxxxx
type=peer
username=SkypeCaller
deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.255
insecure=very
from one of last distro updates , the call from freepbx extension to Skype user is no longer working , asterisk cli says:
Unable to create channel of type ‘SIP’ (cause 20 - Subscriber absent)
I’ve solved this by adding 127.0.0.1/32 as new Local Network in asterisk SIP settings.
Why is this ??
Can I consider it solved or is it only a workaround for a major misconfiguration ??
Thank you