Hi, I have set up my first asterisk system as per O’reillys guide. I have an external SIP provider and have entered their details into the sip.conf. I am testing this out by making calls to my mobile phone and i am getting a few error messages. I have included the details of what i get below along with my configuration. FOr now I have hard coded my own mobile number in the extension just to test it out. It has been blacked out below. Looking at my telecube console it says that this extension is both active and online. I am running this via virtual box. Any help would be greatly appreciated.
What is returned when i call my mobile phone number:(verbose 4; debug 4)
== Using SIP RTP CoS mark 5
-- Executing [0431699654@from-internal:1] Dial("SIP/6001-0000000c", "SIP/@TelecubeLine1") in new stack
== Using SIP RTP CoS mark 5
failed to extend from 64 to 98
failed to extend from 64 to 98
failed to extend from 64 to 98
-- Called SIP/@TelecubeLine1
failed to extend from 64 to 98
failed to extend from 64 to 98
failed to extend from 64 to 98
failed to extend from 64 to 98
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'SIP/6001-0000000c' status is 'CHANUNAVAIL'
My sip.conf
[code][general]
context=default
register = :*@TelecubeLine1
[6001]
type=friend
context=from-internal
host=dynamic
secret=unsecurepassword
disallow=all
allow=ulaw
[TelecubeLine1]
type = peer
host = sip.telecube.net.au
defaultuser = *****
secret = *****
insecure = invite
dtmfmode = rfc2833
disallow = all
allow = ulaw[/code]
pjsip.conf
[code][transport-udp]
type=transport
protocol=udp
bind=0.0.0.0
[6001]
type=endpoint
context=from-internal
disallow=all
allow=ulaw
auth=6001
aors=6001
[6001]
type=auth
auth_type=userpass
password=unsecurepassword
username=6001
[6001]
type=aor
max_contacts=1
[/code]
extentions.conf
[code][from-internal]
exten => 100,1,Answer()
same = n,Wait()
same = n,Playback(hello-world)
same = n,Hangup()
exten => 0431******,1,Dial(SIP/${EXTEND}@TelecubeLine1)[/code]