The sip debugging info is here http://pastebin.com/744005, the sip.conf, extensions.conf and console output are here http://pastebin.com/744065.
 
Thatnk you,
Steven 

Steve Totaro <[EMAIL PROTECTED]> wrote:
We need your dialplan and output from the console to help.

Thanks,
Steve Totaro

Steven Haldeman wrote:
> Hello,
>
> I work for a company that is experimenting with the implementation of
> Asterisk. We have a VoIP provider that is giving us a demo account
> with 200 minutes on it. We can register with their service but cannot
> complete calls with Asterisk. We can use a Grandstream GXP-2000 with
> the supplied registration info and it does work. We can also register
> Asterisk with FWD and dial the FWD loopback and it works. We can dial
> other extensions within Asterisk. The provider said that it appears
> that our usersname is comming over as the extensions after the / in
> the register line.
>
> We are running Asterisk 1.2.6 and the provider is running Tekelec 9000
> switches.
>
> Thanks in advance.
>
> ------------------------------------------------------------------------
> Be a chatter box. Enjoy free PC-to-PC calls
>
> with Yahoo! Messenger with Voice.
> ------------------------------------------------------------------------
>
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