asterisk users - Jan 2005 - Can't initiate a call with X-Lite.

Hello,

	I'm trying to place a call to asterisk using X-Lite. Asterisk is  setup 
with some Grandstream phones. I can call from one grandstream extension 
to another. When I try to an extension with X-Lite, it comes back with 
Status of SIP/2.0 404 Not Found. X-Lite is not registered as asterisk 
extension. It is just sending a sip invite to extension@IP. Does the 
X-Lite need to connect to via a proxy?

After several days of reading RFCs and looking at packet traces, I know 
a bit more about SIP, but not quite enough to make this work.

Is there a way to get asterisk to say what its doing? I tried 
-vvvvvvvvvv etc, but the only messages are see are when I use one of my 
my Grandstream phones. On the wire, is see the same "To:"  header from
both the grandstream and the X-Lite soft phone. I don't understand why 
its "found" by one, and not the other.

Thanks,

	Andy

> 	I'm trying to place a call to asterisk using X-Lite. Asterisk is 
setup
> with some Grandstream phones. I can call from one grandstream extension 
> to another. When I try to an extension with X-Lite, it comes back with 
> Status of SIP/2.0 404 Not Found. X-Lite is not registered as asterisk 
> extension. It is just sending a sip invite to extension@IP. Does the 
> X-Lite need to connect to via a proxy?
No. You should work on configuring xlite to "register" with asterisk.
In the xlite Sip Proxy menu, you will need a "User Name",
"Password",
"Sip Proxy", and "Domain/Realm" defined to match entries in
your
sip.conf definitions.

Your sip.conf for xlite should look something like:
[3005]
type=friend
host=dynamic
username=3005
secret=yourpassword
context=from-sip
canreinvite=no
mailbox=3005 
 
> After several days of reading RFCs and looking at packet traces, I know 
> a bit more about SIP, but not quite enough to make this work.
> 
> Is there a way to get asterisk to say what its doing? I tried 
> -vvvvvvvvvv etc, but the only messages are see are when I use one of my 
> my Grandstream phones. On the wire, is see the same "To:"  header
from
> both the grandstream and the X-Lite soft phone. I don't understand why 
> its "found" by one, and not the other.
>From your asterisk CLI, try "sip debug" to see the flow of packets
to/from
asterisk; "sip no debug" will shut it off.