Help with magic jack and asterisk?

[thecreator]

Hello,
I started working with magic jack trying to get it working.
When I run it, I get:
[Feb 22 14:35:32] NOTICE[9175] pbx_ael.c: AEL load process: compiled config file name '/etc/asterisk/extensions.ael'.
[Feb 22 14:35:32] NOTICE[9175] pbx_ael.c: AEL load process: merged config file name '/etc/asterisk/extensions.ael'.
[Feb 22 14:35:32] NOTICE[9175] pbx_ael.c: AEL load process: verified config file name '/etc/asterisk/extensions.ael'.
[Feb 22 14:35:33] NOTICE[9199] chan_sip.c: Peer 'magicjack' is now Reachable. (576ms / 2000ms)
[Feb 22 14:39:25] NOTICE[9199] chan_sip.c: Call from 'E970462328501' to extension 's' rejected because extension not found.
[Feb 22 14:39:45] NOTICE[9199] chan_sip.c: Call from 'E970462328501' to extension 's' rejected because extension not found.
[Feb 22 14:40:08] NOTICE[9199] chan_sip.c: Call from 'E970462328501' to extension 's' rejected because extension not found.
Any ideas?
Here is what I used for my reference point:
http://www.voipphreak.ca/category/asterisk-tidbits/
If someone has a simple configuration that could be used, that would be great.
I just want to link somehow so I can pick up my handset and call out,, and receive incoming calls, possibly build off that at a later date.
I don't have the time currently to be dealing with this whole thing, My reason in setting this up was so that I could call from anywhere and be able to call out.
I'm in school part of the time, and I can't make long distance calls from their phone, so this was an easy solution.
I also want this to be a stable line; right now it gets unplugged a lot when I move my laptop around.
Thanks,
TheCreator

[mberlant]

I don't have the time currently to be dealing with this whole thing,

Sorry, you don't have any choice but to invest the time and effort in order to reap the reward. You can do that yourself here (or on an Asterisk forum, based upon the scant information you provided), or you can pay someone to do the necessary work for you. But, without the work you won't get the reward.

You have provided precious little to work with here. The one line that says, "rejected because extension not found," indicates that your configuration elements in sip.conf are working and that your Asterisk is Registered with the MJ service. It looks like an incoming call is not being successfully passed from sip.conf to extensions.conf because you have set up sip.conf to send the incoming call to one place (extension s in either a default context or specific context) in extensions.conf and that context (or extension s within that context) does not exist. Your next task is to investigate what piece is missing from this transaction.

As for outgoing calls, you have provided absolutely no information about your symptoms or your configuration, so there is absolutely nothing to go on.