[asterisk-users] Call routing between two Asterisk boxes using SIP not working ...

[Mauro Sergio Ferreira Brasil]

Hello there!

I need some help to configure two Asterix boxes to route calls using SIP.
I followed the instructions present at this site: 
"http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/connecting_two_asterisk.html", 
but I couldn't get it working so far.

The only difference, besides the names that I've used, is that I'm using 
realtime to retrieve all information.

Both boxes registrate on the other perfectly.
The problem happens when one call gets routed. It seems that realtime on 
destination box is trying to find locally a SIP user "1001" that is the 
originator of the call and is a user of the original box.

It finally ends with a: "chan_sip.c:14780 handle_request_invite: Failed 
to authenticate user "1001" <sip:1001 at 10.10.100.158>;tag=as1e79b629" on 
destination box.

Wireshark present on destination box indicates all the following steps:
1- Wengo client registered with user "1001" starts the call to number 
"2001" with Box 1 (at 10.10.100.158);
2- Box 1 makes the challenge;
3- Wengo replies the challenge;
4- Box 1 send an successfull ack to Wengo client and sends the INVITE to 
Box 2 (at 10.10.100.156) that holds user "2001";
5- Box 2 makes the challenge;
6- Box 1 replies the challenge;
7- Box 2 sends a 403 Forbidden;

Has anyone had this problem ?
Can anyone help me out on that ?

Thanks and best regards,

-- 
__At.,                                                                                                                             
   _
 
*Technology and Quality on Information*
Mauro Sérgio Ferreira Brasil
Coordenador de Projetos e Analista de Sistemas
+ mauro.brasil at tqi.com.br <mailto:@tqi.com.br>
: www.tqi.com.br <http://www.tqi.com.br>
( + 55 (34)3291-1700
( + 55 (34)9971-2572