media through NAT problems

<div><div dir="ltr">oh so close.&nbsp; So, my phone is now registering, but media is not connecting(and I do know why but would like verification that this is not fixable????).<br><br>I run wireshark and I see the first few sdps from sjphone to sipx telling it to send media to 192.168...so then I look for the stun setting and then all is well and sipx now sends media to the right ip BUT the wrong port.&nbsp; The sdp in the sip invite is telling<br>
sipx to send the media to port 41598 while the media is coming out of the router at 50123 so naturally, I don't get any media coming back to me here(Is there any fix for this?&nbsp; I mean a stun server can only look at the ip address, by the time sjphone talks to sipx, the port has changed so that info is useless in stun.&nbsp; Is there any way to get sipx to send the media straight back to where it is coming from instead of using the sdp to do so????&nbsp; or is there as I suspect no fix for this and I am screwed and my phone must be on the WAN?<br>
thanks,<br clear="all"><br>-- <br>Dean Hiller<br>CEO/Founder of Extreme Software Offshoring<br><a href="http://xsoftware.biz">http://xsoftware.biz</a><br><a href="http://www.linkedin.com/in/deanhiller">http://www.linkedin.com/in/deanhiller</a><br>
Beijing Cell: 136-991-41547<br>US Phone: 303-376-5776<br><br>
</div></div>

sipx docs missing stuff

<div><div dir="ltr">I finally resolved my problem.&nbsp; The docs on uninstalling on the sipx site were not enough.&nbsp; I found out stuff was being left behind like /etc/sipxpbx and /var/sipxdata.&nbsp; After uninstalling, I wiped these too.&nbsp; It also had something to do with my DNS having sipx as a CNAME aliased to my domain.&nbsp; Uninstalling AND wiping these directories and fixing the DNS to be an A record got my stuff all working.&nbsp; this post is for the next person(hope he finds this needle in a haystack when he needs it).<br><br>now I am trying to figure out why sipx is sending media back to <a href="http://192.168.22.100">192.168.22.100</a> insead of my WAN ip.&nbsp; this should be fun.&nbsp; At least I know it is my final problem and I am set :).<br>
thanks,<br clear="all"><br>-- <br>Dean Hiller<br>CEO/Founder of Extreme Software Offshoring<br><a href="http://xsoftware.biz">http://xsoftware.biz</a><br><a href="http://www.linkedin.com/in/deanhiller">http://www.linkedin.com/in/deanhiller</a><br>
Beijing Cell: 136-991-41547<br>US Phone: 303-376-5776<br><br>
</div></div>

stumped on unauthorized

<div><div dir="ltr">I have setup a sipx system here in my beijing office and I set up one on the wan.&nbsp; The first one was setup with the CD.&nbsp; The second one with "yum install sipxecs".&nbsp; when I use the ip of the first one to register, it works.&nbsp; With the second one, I get 408 unauthorized.&nbsp; The first one naturally has DNS all setup correctly and the second one almost does(My ISP is having technical problems with SRV records deploying to DNS once I put them in their web page...<a href="http://godaddy.com">godaddy.com</a>).<br><br>Anyways have problems like this.&nbsp; How can I debug this problem further?&nbsp; I am stumped.&nbsp; Maybe I should to a complete clean and reinstall and wipe the database, etc. etc.<br>thanks,<br clear="all"><br>-- <br>Dean Hiller<br>
CEO/Founder of Extreme Software Offshoring<br><a href="http://xsoftware.biz">http://xsoftware.biz</a><br><a href="http://www.linkedin.com/in/deanhiller">http://www.linkedin.com/in/deanhiller</a><br>Beijing Cell: 136-991-41547<br>
US Phone: 303-376-5776<br><br>
</div></div>

Re: no firewall on sipx

<div><div dir="ltr">that is what I suspected.&nbsp; <br>thanks,<br>dean<br><br><div class="gmail_quote">On Tue, Aug 19, 2008 at 8:48 AM, Tony Graziano <span dir="ltr">&lt;<a href="mailto:tgraziano <at> myitdepartment.net">tgraziano <at> myitdepartment.net</a>&gt;</span> wrote:<br><blockquote class="gmail_quote">Running iptables will keep sipx from running properly.<br><br>
chkconfig iptables off<br><br>
and turn it off.<br><br>
&gt;&gt;&gt; "Dean Hiller" &lt;<a href="mailto:dean <at> xsoftware.biz">dean <at> xsoftware.biz</a>&gt; 08/18/08 19:53 PM &gt;&gt;&gt;<br><div>
<div></div>
<div class="Wj3C7c">After the CD install of sipx, I looked in /etc/sysconfig and to my<br>
surprise<br>
there was no iptables file and now when I do a service iptables status,<br>
the<br>
firewall is stopped, yet when I do chkconfig --list, it shows the<br>
iptables<br>
should be starting up during boot. &nbsp;I reboot and the firewall is still<br>
stopped(maybe because iptables file is not there). &nbsp;Does anyone have a<br>
good<br>
iptables file I can use and just plop it down on my sipx machine so it<br>
only<br>
allows the upd stuff through and the web pages?<br>
thanks,<br><br>
--<br>
Dean Hiller<br>
CEO/Founder of Extreme Software Offshoring<br><a href="http://xsoftware.biz" target="_blank">http://xsoftware.biz</a><br><a href="http://www.linkedin.com/in/deanhiller" target="_blank">http://www.linkedin.com/in/deanhiller</a><br>
Beijing Cell: 136-991-41547<br>
US Phone: 303-376-5776<br><br>
</div>
</div>
</blockquote>
</div>
<br><br clear="all"><br>-- <br>Dean Hiller<br>CEO/Founder of Extreme Software Offshoring<br><a href="http://xsoftware.biz">http://xsoftware.biz</a><br><a href="http://www.linkedin.com/in/deanhiller">http://www.linkedin.com/in/deanhiller</a><br>
Beijing Cell: 136-991-41547<br>US Phone: 303-376-5776<br><br>
</div></div>

XML-RPC API

<div><div dir="ltr">Hi all,<br><br>I have been given the task to merge sipxecs User portal features to our in house developed ERP Web application written in Python.<br><br>Initially we are looking to have basic functionality such as voicemail Inbox, Follow me configuration on our web app.<br><br>Is XML-RPC the way to go here?<br><br>According to what I have been able to figure out it seems that the SOAP interface is oriented to bulk configuration.<br><br>If indeed XML-RPC is the suggested approach, how can I browse through the API?<br><br>All help and suggestions are appreciated.<br><br>Regards,<br><br>-Luis<br>
</div></div>

Re: DID -> Extension mapping

On Tue, Aug 19, 2008 at 09:53:14AM -0300, Tony Graziano wrote:
> Ranga: Do you have a "how-to" for bandtel yet? Can sipXbridge strip the digits
> instead of using a dial plan entry?
> 

It would be nice to have sipXbridge strip the 0300 and 0200 (for toll free DIDs only, afaik) off iff the ITSP is
Bandtel.  I'll gladly document my experience getting it working on the sipx wiki, too.

> We have a bandtel trunk here, but are using an ingate. We strip the 0300 prefix
> and deliver the calls as 10 digits as an alias to the sipx system.
>  
> your alias should be "1234567890" (no "domain.tld". You should try to make sure
> there is nothing wrong with the dial plan by disabling it and making
> 03001234567890 an alias to, then seeing if the call completes, this way it
> tells you they are delivering the call. I do not believe a dial plan entry
> would be necessary 9at least it wasnt necessary in 3.10.2) to achive this.
>  
> If when looking at your call logs, you have see the invite from bandtel as
> DID <at> ipaddress, then make sure you have an domain alias to the IP (remember to
> apply and activate any alias or dial plan changes).
>  

That IP was a domain alias in my mind.  I just disabled the dialplan, added the 0300<DID> alias to a user, and
added the public IP as a domain alias.  That was the missing piece - it's working as I would expect now.  

My dialplan to strip the 0300 is also working now, too.  

Thanks VERY much.

> >>> Richard Kolkovich <sarumont <at> sigil.org> 08/19/08 08:02AM >>>
> On Mon, Aug 18, 2008 at 11:18:05AM -0500, Richard Kolkovich wrote:
> > I'm setting up a sipx installation using Bandtel as a SIP trunk.  For the
> office, I need to have DIDs map to extensions.  When I get the INVITE from
> Bandtel, the number is prepended with 0300 (i.e. - 03001234567890@).  I have
> created a DialPlan based on http://tinyurl.com/5wjmtb that sets the prefix to
> 0300, matching 10 digits.  The resulting call is no prefix and matched suffix. 
> This should, as I understand it, result in a call to 1234567890.
> 

-- 

Richard Kolkovich
sarumont <at> sigil.org

On Tue, Aug 19, 2008 at 09:53:14AM -0300, Tony Graziano wrote:
> Ranga: Do you have a "how-to" for bandtel yet? Can sipXbridge strip the digits
> instead of using a dial plan entry?
> 

It would be nice to have sipXbridge strip the 0300 and 0200 (for toll free DIDs only, afaik) off iff the ITSP is
Bandtel.  I'll gladly document my experience getting it working on the sipx wiki, too.

> We have a bandtel trunk here, but are using an ingate. We strip the 0300 prefix
> and deliver the calls as 10 digits as an alias to the sipx system.
>  
> your alias should be "1234567890" (no "domain.tld". You should try to make sure
> there is nothing wrong with the dial plan by disabling it and making
> 03001234567890 an alias to, then seeing if the call completes, this way it
> tells you they are delivering the call. I do not believe a dial plan entry
> would be necessary 9at least it wasnt necessary in 3.10.2) to achive this.
>  
> If when looking at your call logs, you have see the invite from bandtel as
> DID <at> ipaddress, then make sure you have an domain alias to the IP (remember to
> apply and activate any alias or dial plan changes).
>  

That IP was a domain alias in my mind.  I just disabled the dialplan, added the 0300<DID> alias to a user, and
added the public IP as a domain alias.  That was the missing piece - it's working as I would expect now.  

My dialplan to strip the 0300 is also working now, too.  

Thanks VERY much.

> >>> Richard Kolkovich <sarumont <at> sigil.org> 08/19/08 08:02AM >>>
> On Mon, Aug 18, 2008 at 11:18:05AM -0500, Richard Kolkovich wrote:
> > I'm setting up a sipx installation using Bandtel as a SIP trunk.  For the
> office, I need to have DIDs map to extensions.  When I get the INVITE from
> Bandtel, the number is prepended with 0300 (i.e. - 03001234567890@).  I have
> created a DialPlan based on http://tinyurl.com/5wjmtb that sets the prefix to
> 0300, matching 10 digits.  The resulting call is no prefix and matched suffix. 
> This should, as I understand it, result in a call to 1234567890.
> 

--

-- 

Richard Kolkovich
sarumont <at> sigil.org

Asterisk -siPX-OCS

<div>

<div class="Section1">

<p class="MsoNormal"><span lang="EN-US">Hi,<p></p></span></p>

<p class="MsoNormal"><span lang="EN-US"><p>&nbsp;</p></span></p>

<p class="MsoNormal"><span lang="EN-US">Any one has success integrate
Asterisk-sipX-OCS?<p></p></span></p>

<p class="MsoNormal"><span lang="EN-US"><p>&nbsp;</p></span></p>

<p class="MsoNormal"><span lang="EN-US">SInjo<p></p></span></p>

</div>

</div>

no firewall on sipx

<div><div dir="ltr">After the CD install of sipx, I looked in /etc/sysconfig and to my surprise there was no iptables file and now when I do a service iptables status, the firewall is stopped, yet when I do chkconfig --list, it shows the iptables should be starting up during boot.&nbsp; I reboot and the firewall is still stopped(maybe because iptables file is not there).&nbsp; Does anyone have a good iptables file I can use and just plop it down on my sipx machine so it only allows the upd stuff through and the web pages?<br>
thanks,<br clear="all"><br>-- <br>Dean Hiller<br>CEO/Founder of Extreme Software Offshoring<br><a href="http://xsoftware.biz">http://xsoftware.biz</a><br><a href="http://www.linkedin.com/in/deanhiller">http://www.linkedin.com/in/deanhiller</a><br>
Beijing Cell: 136-991-41547<br>US Phone: 303-376-5776<br><br>
</div></div>

DID -> Extension mapping

I'm setting up a sipx installation using Bandtel as a SIP trunk.  For the office, I need to have DIDs map to
extensions.  When I get the INVITE from Bandtel, the number is prepended with 0300 (i.e. -
03001234567890@).  I have created a DialPlan based on http://tinyurl.com/5wjmtb that sets the prefix to
0300, matching 10 digits.  The resulting call is no prefix and matched suffix.  This should, as I understand
it, result in a call to 1234567890.

The extension I want to forward to has the DID as an alias, so I would assume this dial plan would ring that
extension.  It actually results in a 404 sent back to Bandtel, and the call failing.

Is it possible to accomplish this without making the extensions the DIDs themselves?

Thanks,

-- 

Richard Kolkovich
sarumont <at> sigil.org

I'm setting up a sipx installation using Bandtel as a SIP trunk.  For the office, I need to have DIDs map to
extensions.  When I get the INVITE from Bandtel, the number is prepended with 0300 (i.e. -
03001234567890@).  I have created a DialPlan based on http://tinyurl.com/5wjmtb that sets the prefix to
0300, matching 10 digits.  The resulting call is no prefix and matched suffix.  This should, as I understand
it, result in a call to 1234567890.

The extension I want to forward to has the DID as an alias, so I would assume this dial plan would ring that
extension.  It actually results in a 404 sent back to Bandtel, and the call failing.

Is it possible to accomplish this without making the extensions the DIDs themselves?

Thanks,

--

-- 

Richard Kolkovich
sarumont <at> sigil.org

variables in phone group configurations

<div>Hi all,<div><br></div>
<div>Is it possible to use variables in phone group configurations that would be interpreted before creating the config files for phones?</div>
<div><br></div>
<div>For example, we use Linksys SPA942 phones and I would like the following configs:</div>
<div><br></div>
<div>Under Phones, Phone, Station Name I would like to be able to use something like $username</div>
<div>
<div>Under Lines,&nbsp;Call Feature Settings,&nbsp;Voice Mail Server I would like to set&nbsp;<a href="mailto:%24extension <at> voip.domain">$extension <at> voip.domain</a>
</div>
<div><br></div>
</div>
<div>The first one I can set manually under each phones configuration and the second under each lines configuration for the phone but this is relatively cumbersome whereas a set of variables that can be replaced with appropriate values while creating the phone profiles would be much quicker and less error prone.</div>
<div><br></div>
<div>With luck something like this is already available and I've just missed the obvious.</div>
<div><br></div>
<div>Regards,</div>
<div>
<br><div> <span class="Apple-style-span"><div><span class="Apple-style-span"><div><span class="Apple-style-span"><div><span class="Apple-style-span"><div><span class="Apple-style-span"><div><span class="Apple-style-span"><div><span class="Apple-style-span"><div><div><span class="Apple-style-span"><span class="Apple-style-span">Andrew Radke<br>Yuruga Nursery Pty Ltd<br>Clonal Solutions Australia Pty Ltd<br>PO Box 220<br>Walkamin Qld 4872<br>Phone: (07) 4093 3826<br>Fax: (07) 4093 3869<br>Email:&nbsp;<a href="file:///Network/Servers/mac-server.internal.yuruga.com.au/Users/andrew.radke/Application%20Data/Microsoft/Signatures/andrew.radke <at> yuruga.com.au">andrew.radke <at> yuruga.com.au</a><br>Web:&nbsp;<a href="http://www.yuruga.com.au/">www.yuruga.com.au</a></span></span></div></div></span></div></span></div></span></div></span></div></span></div></span></div></span> </div>
<br>
</div>
</div>

OCS-sipX

<div>

<div class="Section1">

<p class="MsoNormal"><span lang="EN-US">Hello,<p></p></span></p>

<p class="MsoNormal"><span lang="EN-US"><p>&nbsp;</p></span></p>

<p class="MsoNormal"><span lang="EN-US"><p>&nbsp;</p></span></p>

<p class="MsoNormal"><span lang="EN-US">I am integrating OCS and sipX. I have a
problem which the calls both direction can not established/not ringing at both
end-point.<p></p></span></p>

<p class="MsoListParagraph"><span lang="EN-US"><span>1.<span>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><span lang="EN-US">OCS to sipX. When calling sipx
to OCS below the debug I capture on Mediation. Supposed TO is to sipX which the
IP <at>  172.18.1.211 or sipX.eltech.com (sipX FQDN) :<p></p></span></p>

<p class="MsoListParagraph"><span lang="EN-US"><p>&nbsp;</p></span></p>

<p class="MsoListParagraph"><span lang="EN-US">&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;&lt;Incoming
RawDataBuffer 172.18.1.200:5061&lt;-172.18.1.210:4345<p></p></span></p>

<p class="MsoListParagraph"><span lang="EN-US">

eINVITE sip:+3345 <at> mediation.esi.com:5061;user=phone;maddr=mediation.esi.com
SIP/2.0<p></p></span></p>

<p class="MsoListParagraph"><span lang="EN-US">ms-user-data:
ms-publiccloud=false;ms-federation=false<p></p></span></p>

<p class="MsoListParagraph"><span lang="EN-US">Record-Route:
&lt;sip:ocs.esi.com:5061;transport=tls;ms-role-rs-to;lr&gt;;tag=E2C492B1BFDBD373E90670F181727F67<p></p></span></p>

<p class="MsoListParagraph"><span lang="EN-US">Via: SIP/2.0/TLS 172.18.1.210:4345;branch=z9hG4bKC4428C1B.04D62948;branched=TRUE<p></p></span></p>

<p class="MsoListParagraph"><span lang="EN-US">Max-Forwards: 69<p></p></span></p>

<p class="MsoListParagraph"><span lang="EN-US">Content-Length: 1071<p></p></span></p>

<p class="MsoListParagraph"><span lang="EN-US">Via: SIP/2.0/TLS
127.0.0.1:50071;received=172.18.0.205;ms-received-port=50072;ms-received-cid=4100<p></p></span></p>

<p class="MsoListParagraph"><span lang="EN-US">P-Asserted-Identity:
"sinjo"&lt;sip:sinjo <at> esi.com&gt;,&lt;tel:+2179751234&gt;<p></p></span></p>

<p class="MsoListParagraph"><span lang="EN-US">From:
"sinjo"&lt;sip:sinjo <at> esi.com&gt;;tag=b5b8cf48d8;epid=75094f81c7<p></p></span></p>

<p class="MsoListParagraph"><span lang="EN-US">To:
&lt;sip:+3345 <at> esi.com;user=phone&gt;<p></p></span></p>

<p class="MsoListParagraph"><span lang="EN-US">Call-ID:
181afa82020049be9e8b430e2d924357<p></p></span></p>

<p class="MsoListParagraph"><span lang="EN-US">CSeq: 1 INVITE<p></p></span></p>

<p class="MsoListParagraph"><span lang="EN-US">Contact:
&lt;sip:sinjo <at> esi.com;opaque=user:epid:wLR4oI10klSGm5O0NMbKegAA;gruu&gt;<p></p></span></p>

<p class="MsoListParagraph"><span lang="EN-US">User-Agent: UCCP/2.0.6362.64
OC/2.0.6362.64 (Microsoft Office Communicator)<p></p></span></p>

<p class="MsoListParagraph"><span lang="EN-US">Ms-Conversation-ID:
Acj+yVOhS+hypfSFRuWyTv8Qmzq8OQ==<p></p></span></p>

<p class="MsoListParagraph"><span lang="EN-US">Supported: timer<p></p></span></p>

<p class="MsoListParagraph"><span lang="EN-US">Supported: ms-sender<p></p></span></p>

<p class="MsoListParagraph"><span lang="EN-US">Supported: ms-early-media<p></p></span></p>

<p class="MsoListParagraph"><span lang="EN-US">ms-keep-alive: UAC;hop-hop=yes<p></p></span></p>

<p class="MsoListParagraph"><span lang="EN-US">Supported: ms-conf-invite<p></p></span></p>

<p class="MsoListParagraph"><span lang="EN-US">Content-Type: application/sdp<p></p></span></p>

<p class="MsoListParagraph"><span lang="EN-US"><p>&nbsp;</p></span></p>

<p class="MsoListParagraph"><span lang="EN-US"><p>&nbsp;</p></span></p>

<p class="MsoListParagraph"><span lang="EN-US"><p>&nbsp;</p></span></p>

<p class="MsoListParagraph"><span lang="EN-US"><p>&nbsp;</p></span></p>

<p class="MsoListParagraph"><span lang="EN-US"><p>&nbsp;</p></span></p>

<p class="MsoListParagraph"><span lang="EN-US"><span>2.<span>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><span lang="EN-US">sipX to OCS. Below the debug
captured on Mediation. Supposed the TO no to IP <at>  is 172.18.1.211 because this
is the IP <at>  of sipX.<p></p></span></p>

<p class="MsoListParagraph"><span lang="EN-US"><p>&nbsp;</p></span></p>

<p class="MsoListParagraph"><span lang="EN-US">&gt;&gt;&gt; Incoming TCP packet
BEGIN<p></p></span></p>

<p class="MsoListParagraph"><span lang="EN-US">INVITE
sip:2179751234 <at> 172.18.1.201:5060;transport=tcp SIP/2.0\r\n<p></p></span></p>

<p class="MsoListParagraph"><span lang="EN-US">Record-Route:
&lt;sip:172.18.1.211:5060;lr;sipXecs-rs=%2Afrom%7EMmEyOWFmNGY%60.400_authrules%2Aauth%7E%2162baf8ecf030d04ea30f350c1d357be1&gt;\r\n<p></p></span></p>

<p class="MsoListParagraph"><span lang="EN-US">Max-Forwards: 16\r\n<p></p></span></p>

<p class="MsoListParagraph"><span lang="EN-US">Contact:
&lt;sip:3345 <at> 172.18.0.205:63498&gt;\r\n<p></p></span></p>

<p class="MsoListParagraph"><span lang="EN-US">To:
"02179751234"&lt;sip:02179751234 <at> 172.18.1.211&gt;\r\n<p></p></span></p>

<p class="MsoListParagraph"><span lang="EN-US">From:
"Sinjo"&lt;sip:3345 <at> 172.18.1.211&gt;;tag=2a29af4f\r\n<p></p></span></p>

<p class="MsoListParagraph"><span lang="EN-US">Call-Id:
OWQxNWMxZjgzNGRlNmQ3YzRhY2QzNzVkMDk3ZjEzNmY.\r\n<p></p></span></p>

<p class="MsoListParagraph"><span lang="EN-US">Cseq: 1 INVITE\r\n<p></p></span></p>

<p class="MsoListParagraph"><span lang="EN-US">Allow: INVITE, ACK, CANCEL, OPTIONS,
BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO\r\n<p></p></span></p>

<p class="MsoListParagraph"><span lang="EN-US">Content-Type: application/sdp\r\n<p></p></span></p>

<p class="MsoListParagraph"><span lang="EN-US">User-Agent: X-Lite release 1011s
stamp 41150\r\n<p></p></span></p>

<p class="MsoListParagraph"><span lang="EN-US">Content-Length: 233\r\n<p></p></span></p>

<p class="MsoListParagraph"><span lang="EN-US">Date: Fri, 15 Aug 2008 11:22:27
GMT\r\n<p></p></span></p>

<p class="MsoListParagraph"><span lang="EN-US">Via: SIP/2.0/TCP 172.18.1.211;branch=z9hG4bK-sipXecs-000d219b4b79f93ba9f5705313e1de232fe3\r\n<p></p></span></p>

<p class="MsoListParagraph"><span lang="EN-US">Via: SIP/2.0/UDP
172.18.1.211;branch=z9hG4bK-sipXecs-000a2b88e2de789613c1a6057211d05d22dd~9bdd594f42281f01499ac7643eefdf5f\r\n<p></p></span></p>

<p class="MsoListParagraph"><span lang="EN-US">Via: SIP/2.0/UDP
172.18.1.211;branch=z9hG4bK-sipXecs-00059796ec64a36373cb15b04bf85d37e782~00f377e4a4073705fd337d222de50ac9\r\n<p></p></span></p>

<p class="MsoListParagraph"><span lang="EN-US">Via: SIP/2.0/UDP
172.18.0.205:63498;branch=z9hG4bK-d87543-ea5af023e70f7e29-1--d87543-;rport=63498\r\n<p></p></span></p>

<p class="MsoListParagraph"><span lang="EN-US">\r\n<p></p></span></p>

<p class="MsoNormal"><span lang="EN-US"><p>&nbsp;</p></span></p>

<p class="MsoNormal"><span lang="EN-US"><p>&nbsp;</p></span></p>

<p class="MsoNormal"><span lang="EN-US">Any one can help how to fix?<p></p></span></p>

<p class="MsoNormal"><span lang="EN-US"><p>&nbsp;</p></span></p>

<p class="MsoNormal"><span lang="EN-US">Tks.<p></p></span></p>

<p class="MsoNormal"><span lang="EN-US">Sinjo<p></p></span></p>

</div>

</div>