Hi all.
Ive recently setup two Asterisk boxes (running Asterisk@Home to be specific),
and Im trying to get a trunk going between them.
So far I have tried a combination of IAX and SIP configuring them through AMP
and writing the config files manually, but I cant seem to get calls going
between the two.
I have named each box asterisk1 and asterisk2.
Does anyone have some working SIP and/or IAX trunk configurations they can send
to me?
Here is my current SIP config which doesnt seem to work:
sip.conf on asterisk1:
register=ast1:****@x.x.x.x
[100]
username=100
type=friend
secret=****
record_out=Never
record_in=Never
qualify=no
port=5060
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid="Ext 100" <100>
[ast2]
type=user
secret=****
context=local
[astrx2]
username=ast1
type=peer
secret=****
host=x.x.x.x
sip.conf on asterisk2:
register=ast2:****@y.y.y.y
[101]
username=101
type=friend
secret=****
record_out=Never
record_in=Never
qualify=no
port=5060
nat=never
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid="Ext 101" <101>
[ast1]
type=user
secret=****
context=local
[astrx1]
username=ast2
type=peer
secret=****
host=y.y.y.y
Where x.x.x.x and y.y.y.y are the IP addresses of each opposite box respectively
(I'd put the real IPs in there but they are public routable ones :-) )
Basically what I get is the following:
If I dial 43100 from ext 101 on asterisk2, this should transfer the call to
asterisk1 and ring ext 100 on asterisk1. All I get is "all circuits are
busy".
If I dial 44101 from ext 100 on asterisk 1, it should send the call to
asterisk2, but I get absolutely nothing. Not an error tone or even a ringing
sound.
When I look at Asterisk Info via AMP this is what I see under SIP peers:
on asterisk1:
x.x.x.x:5060 ast1 120 Request Sent
on asterisk2:
y.y.y.y:5060 ast2 120 Unregistered
Does anyone have any ideas as to why or what might be happening and how I can
fix it?
Cheers,
Tom
