Help in sip.config and extension.conf

Hi,

I have configured the my asterisk for testing purpose i have configured sip.conf and extensions.conf. at client end i am using SJphone software on two PC.
I am unable to make a to other SJphone. when i am calling from one SJphone <1001> SJphone is open the director.

Please see the configuration which i have done.
is any thing is missing?

sip.conf

port=5060
bindaddr=0.0.0.0
srvlookup=no

[1000]
context=fromsip
type=friend
;username=1000
;secret=1000
host=10.0.0.204
nat=no
canreinvite=no
disallow=all
allow=ulaw
qualify=1000

[1001]
context=fromsip
type=friend
;username=1001
;secret=1001
callerid="ME"
host=10.200.0.14
nat=no
canreinvite=no
disallow=all
allow=ulaw
qualify=1000

extensions.conf

exten => 1000,1,Dial(SIP/1000,20,tr)
exten => 1001,1,Dial(SIP/1001,20,tr)

when i am executing sip show peers command then i am getting following result

*CLI> sip show peers
Name/username Host Dyn Nat ACL Mask Port Status
1002/1002 (Unspecified) D N 255.255.255.255 0 UNKNOWN
1001 10.200.0.14 255.255.255.255 5060 OK (1 ms)
1000 10.0.0.204 255.255.255.255 5060 OK (1 ms)
phone2 10.200.0.11 255.255.255.255 5060 Unmonitored
*CLI>

Can both phones talk to Asterisk? Have you tried Asterisk’s echo test with both of them, for example?

The first thing to do is to get each phone to talk to Asterisk. Then you can try and get them to talk to each other through Asterisk. Don’t try and get two phones working before you know they can each communicate properly with Asterisk.