ask for help

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I am confused about a problem with pjsua. 
   I built a sip server with Brekeke ondo sip server on my computer .Its IP is 218.9.121.185 .I can register to server successfully on my computer.
Below is my way to register
>>> +a
Your SIP URL: (empty to cancel): sip:ljm at 218.9.121.185
URL of the registrar: (empty to cancel): sip:2189.121.185
Auth Realm: (empty to cancel): *
Auth Username: (empty to cancel): ljm
Auth Password: (empty to cancel): 123
 
Below is register information on the server:
User Contact URL Detail 
ljm 
 sip:ljm at 218.9.121.185:6000 Expires :  300 
Priority :  1000 
Accept Pattern :    
Requester :  218.9.121.185:6000 
Time Update : Wed Mar 18 11:58:12 CST 2009 
 
Below is another one who registers to server:
>>> +a
Your SIP URL: (empty to cancel): sip:zj at 218.9.124.236
URL of the registrar: (empty to cancel): sip:218.9.121.185
Auth Realm: (empty to cancel): *
Auth Username: (empty to cancel): zj
Auth Password: (empty to cancel): 123
He also can register to server successfully,below is his register information on the server:
zj 
 sip:zj at 218.9.124.236:6000 Expires :  300 
Priority :  1000 
Accept Pattern :    
Requester :  218.9.124.236:6000 
Time Update : Wed Mar 18 12:40:06 CST 2009 
 
He can call me successfully,and I can hear ringing,below is his way to call:
>>> m
(You currently have 0 calls)
Buddy list:
 -none-
Choices:
   0         For current dialog.
  -1         All 0 buddies in buddy list
  [1 - 0]    Select from buddy list
  URL        An URL
  <Enter>    Empty input (or 'q') to cancel
Make call: sip:ljm at 218.9.121.185
Then I will get message and I will hear ringing:
You have 1 active call
Current call id=0 to <sip:zj at 218.9.124.236> [INCOMING]
>>>  12:42:10.984     ec00AF6BE8  Underflow, buf_cnt=0, will generate 1 frame
 
But I can't call him,he can't hear ringing:
12:49:42.187    pjsua_app.c  Call 2 is DISCONNECTED [reason=408 (Request Timeou
)]
12:49:42.203    pjsua_app.c
 [DISCONNCTD] To: sip:zj at 218.9.124.236
   Call time: 00h:00m:00s, 1st res in 32719 ms, conn in 0ms
   SRTP status: Not active Crypto-suite: (null)
 
I found a problem according to testing that if I register on my computer (it also is running sip server ),everyone who registered can call me successfully ,but I can't call back. A and B(no one is on my computer) can register to the server,but they can't call each other.
I don't know the reason. what's wrong with it? thank you !


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