Sip trunking issue

Asterisk Asterisk Support
1

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 ??

2

No [1000] in sip.conf

3

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

4

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.