Hi. My name is Phil and I am a noob. (got that out of the way)
FreePBX 3.2.11
Asterisk (Ver. 1.8.22.0)
I have put together a SIP only FreePBX box. I am using soft phones on the desktop. Phones register with server. Extensions call extensions. I setup an IVR for Inbound Route. Tested and works.
I’m getting the default congestion message when doing outbound calls.
I have labored through the trunk setup and finally put together a setup that actually registers the trunk. Here are the trunk settings:
peer settings:
disallow=all
allow=ulaw
username=(phonenumber)
fromuser=(phonenumber)
type=peer
secret=(password)
qualify=yes
maxexpirey=3600
host=(provider)
fromdomain=(provider)
insecure=very
dtmfmode=auto
session-timers=refuse
defaultexpirey=60
nat=yes
canreinvite=no
context=from-trunk
user settings:
authname=(phonenumber)
canreinvite=no
context=from-pstn
dtmf=inband
dtmfmode=inband
fromdomain=(provider)
fromuser=(phonenumber)
host=(provider)
insecure=very
secret=(password)
type=user
user=phone
username=(phonenumber)
Registration String:
(phonenumber):(password)@(provider)/(phonenumber)
I have this message in the logs when making an outbound call:
WARNING[19588] app_dial.c: Unable to create channel of type ‘SIP’ (cause 20 - Subscriber absent)
I’m stuck. Can somebody help me?
Thanks,