Sip trunking issue

hello,
i m trying to connect two asterisk serever by sip trunking. i m using soekris box as one of the asterisk server with Astlinux installed and i m using redhat interprise linux 5 on the other server .i have used the basic example given in the book ‘asterisk the future of telephony’ . now when i have completed all the changes in the dialplan of both the servers. i am able to route calls to rhel 5 server from soekris server but there is some error in calls routing from rhel 5 server to soekris server. the messege displayed on CLI is :

== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
– Executing [1000@phones:1] NoOp(“SIP/1001-0000002a”, “”) in new stack
– Executing [1000@phones:2] Dial(“SIP/1001-0000002a”, “SIP/1000,30”) in new stack
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
[Aug 6 09:51:53] WARNING[16146]: chan_sip.c:5618 create_addr: No such host: 1000
[Aug 6 09:51:53] WARNING[16146]: app_dial.c:1780 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 20 - Unknown)

is there some problem with the drivers here ?? or i m writing the dialplan incorrectly ??

No [1000] in sip.conf

but it is registered in sip.conf file.??

What do you mean by “registered”? Normally you should use static addresses, rather than register, for point to point “trunks”. In all cases you need a sip.conf section.

The error indicates that it failed to find a sip.conf section name of 1000, so it is interpreting the 1000 as a domain name, and also failing to find that.

I suppose it is just possible that it is confusing and is finding the section, but the host name in that section is unresolvable.