Can't Make Calls

1. Brekeke Product Name and version: Sip Server Version 2.4.3.9 Evaluation

2. Java version: ver 6

3. OS type and the version: Windows server 2003

4. UA (phone), gateway or other hardware/software involved: X-Lite, Grandstream 286

5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/ ... terns.html : Pattern 3

6. Your problem:
Sip Server is installed on a computer running win server 2003, and x-lite is installed on the same machine, router used in Thomson 585v7, firewall is disabled, port mapping for 5060 and UDP port range 10000-10999 is done to the ip of the machine which is running the brekeke sip server. On another network I have 2 grandstream handytone 286 devices behind the same model of router and also firewall is disabled. I can make calls from one handytone to the x-lite running on the server but when I try to call the other handytone I can see the session active on the sip server but the other phone doesn't ring!!! what could be wrong?

Please help as I am testing the software and i need to get it to work before purchase.

thanks

From packets, is there any sip response code sent back from Grandstream handytone?
are the handytone side phones registered at Brekeke sip server?
or there is any dial plan?
if capture packets at handytone side, is there INVITE coming in?

Yes the handytones and the x-lite are registered at the sip server, also when I make a call from one hanytone to another I see at the log that user 110 is inviting user 160, but I never hear a ring on the other side.

There is no dial plan, I don't know what the dial plan does?

This is the Session Details when I make a call from user 160 to user 110.

EX-SID 84
From-uri sip:160@196.219.160.162 [behind NAT]
From-ip 196.219.160.225:65119 (UDP)
From-if 196.219.160.162:5060
To-uri sip:110@196.219.160.162:5060 [behind NAT]
To-ip 196.219.160.225:49264 (UDP)
To-if 196.219.160.162:5060
Call-ID 6a5b8a9a10746d2b@192.168.1.160
rule registered=sip:110(sip:110@192.168.1.200)
plug-in InviteSession
sip-packet-total 1
listen-port 5060
session-status Inviting
time-inviting Fri May 07 21:46:43 EEST 2010
rtp-relay on
rtp-srcdst
rtp-dstsrc
media audio
transport RTP/AVP
payload -
status active
listen-port 10000
send-port
target 196.219.160.225:5004
packet-count 0
packet/sec 0
current size 0
buffer size 260

change configuration tab/sip/NAT traversal/Interval(ms) to 10000 or smaller number and restart sip server

I changed it to 9999, restarted the Sip server tried to make several calls but still it doesn't work.

Please advice what else could be done?

if making a call from xlite to one of handytone, does the handytone side ring?
use the following dial plan to make handytones registration time shorter.
matching
$request = ^REGISTER
$addr = handytone_globalip //196.219.160.225

deploy
$continue=true
&register.contact.remote=true
&net.registrar.adjust.expires = 6

save the dial plan and click "apply Rules" button.

if still not work, try another router at handytone side.