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<title>IP (VoIP) to GSM Gateway 'Blue Voice' Internet Calling Siemens PBX by 2N
VoIP Gateway</title>
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<h1 align="center">The Smartest Way from IP to GSM Networks -</h1>
<h1 align="center"><font size="5">VoiceBlue Lite & Enterprise
VoIP GSM Gateway</font></h1>
<hr>
<p>Voice Blue Lite - the first professionally built VoIP GSM
Gateway! It is an ideal complementary product to any SIP-based
IP PBX. It allows the user to get from the IP to a GSM network
and vice versa very cheaply. The Ethernet (RJ 45) connection is
a significant advantage in most cases. </p>
<h2>What is Voice Blue Lite?</h2>
<p>The Voice Blue Lite GSM Gateway is an ideal complementary
product to any SIP-based IP PBX. It's suitable for small and
medium companies with IP infrastructure and for companies with
international affiliates. With Voice Blue Lite you gain
significant savings on outgoing and incoming calls from IP to
GSM networks and backwards. Thanks to the efficient and powerful
Least Cost Router (LCR), the Voice Blue Lite GSM gateway always
chooses the cheapest possible way to route the call (according
to GSM prefixes, free minutes on SIM cards etc.). Using the
Voice Blue Lite GSM Gateway you achieve complete independence
when connecting to GSM networks and maintain 100% control of
your GSM call costs. With the help of voice prompts and
efficient Dynamic Clip Routing, Voice Blue Lite routes incoming
GSM calls to the right IP phones.<br>
<br>
<strong>Voice Blue Lite offers VoIP services</strong>( internet
telephony), all functions and advantages of digital GSM Gateways
and many other features. Thanks to its efficient LCR, the Voice
Blue Lite VoIP GSM Gateway (Fixed Cellular Terminal) always
chooses the cheapest route to any GSM network used. Gaining a
100%control of all outgoing GSM calls is one of the main
advantages of our VoIP GSM Gateway compared with other,
"home-made" solutions. With the help of voice prompts and
efficient CLIP routing, <strong>Voice Blue Lite VoIP GSM Gateway
always routes GSM calls to the right IP phone.</strong> <br>
By connecting Voice Blue Lite to your corporate (VoIP company),
you can use SMS as an additional communication tool with your
clients. SMS delivery reports are obtained automatically.<a rel="nofollow" target="_blank" href="http://www.2n.cz/news/press_releases/faq-voip-gsm-gateway-voice-blue-lite.html?"><br>
</a><br>
<strong>Voice Blue Lite VoIP GSM Gateway (Fixed Cellular
Terminal) has been successfully tested and proved as compatible
with following systems:</strong> </p>
<ul>
<li>Cisco Call Manager</li>
<li>Alcatel Omni PCx Enterprise systems </li>
<li>Siemens Hi Path 2 000</li>
<li>Asterisk PBX system</li>
<li>OnDo IP PBX</li>
<li>Televentage equipment</li>
<li>Epygi IP PBX</li>
<li>Call Manager Express</li>
<li>And other</li>
</ul>
<h3>NEW FEATURE:</h3>
<p><strong>Mobility Extension</strong>- makes you accessible any
time and anywhere on a mobile phone due to the Follow Me
function and allows to transfer call back to your PBX extension
due to a Mobility Extension function</p>
<h3>Features:</h3>
<ul>
<li>Up to four GSM channel gateway<img src="http://www.2n.cz/images2/obrazek_click/151/thumbnail.jpg" class="hand" align="right" border="0"></li>
<li>Mobility Extension<br>
</li>
<li>Standard SIP client</li>
<li>Smart Voice Routing - Least Cost Routing (LCR)</li>
<li>Intelligent incoming call routing</li>
<li>SMS server - support for SMS messages sending &
receiving</li>
<li>LCR according to free minutes</li>
<li>Internal Callback</li>
<li>SMS Callback support (XAPI server)<br>
</li>
<li>Top voice quality (EFR super sound)</li>
<li>Worldwide use (GSM 900/1800 MHz and 850/1900MHz)</li>
<li>DISA with voice navigation, user defined voice message
for incoming calls<img src="http://www.2n.cz/images2/obrazek_click/152/thumbnail.jpg" class="hand" align="right" border="0"></li>
<li>AT command based administration</li>
<li>CDR buffer for up to 500,000 records, Compact flash</li>
<li>LOG and statistics saving</li>
<li>Integrated antenna splitter</li>
</ul>
<h1>The Most Effective Way from IP to GSM</h1>
<hr>
<p><strong><br>
VoiceBlue Enterprise is a VoIP GSM Gateway supporting
SIP/H.323protocols with SIP Proxy. It is a complementary product
to any SIP/H.323 based IP PBX and can also be used as a
substitute to any SIP based
<a rel="nofollow" target="_blank" href="http://www.2n.cz/products/pbx/omega_pbx.html?">
IP PBX</a> because the SIP proxy server can be included.</strong></p>
<p>The Voice Blue Enterprise fixed cellular terminal can be used
not only as a gateway to GSM but also as a SIP based IP PBX.<strong>
<br>
</strong>As Voice Blue Enterprise uses the Siemens MC55/56
modules, it is capable of data transfer (GPRS technology is
implemented) and therefore can serve as a gateway from your LAN
to the Internet. A firewall is embedded to provide maximum
security. <br>
</p>
<p>All these features can be active and physically used at the
same time. The whole system can be supplied as a "puzzle", which
means that the customer chooses only the features he wants to
have in the gateway. <strong>The customer shall pay no money for
the features that he doesn't need. </strong>With the latest
version of web based SMS server you will gain the advantage of a
full review of your messages, whether they were delivered and
exactly when.<br>
----------------------------------------------------</p>
<div align="left">
<br>
</div>
<div align="left">
<strong>The VoiceBlue Enterprise Fixed Cellular Terminal has
been successfully tested and proved as compatible with
following systems:</strong> </div>
<div align="left">
<ul>
<li>Cisco Call Manager</li>
<li>Alcatel Omni PCx Enterprise systems </li>
<li>Siemens Hi Path 3 000, 3 500, 3 800 PBX system</li>
<li>Asterisk PBX system <br>
</li>
<li>OnDo IP PBX</li>
<li>Televentage equipment</li>
</ul>
</div>
<h4 align="center"><font size="5">Flash animated case studies:</font></h4>
<p><strong>
<a rel="nofollow" target="_blank" href="http://www.2n.cz/popup/vbe-hotel.html?">
<img src="http://www.2n.cz/images2/obrazek/294/onpage.gif" alt="Click to play 2N VoiceBlue Enterprise - Simple IP PBX" align="left">Simple
IP PBX</a></strong></p>
<p>VoiceBlue Enterprise provides calls from the IP network to
the GSM and interconnection with the IP network using the SIP,
which enables you to make free calls to other VoIP users over
the Internet or use the VoIP providers' services as well. It can
be used as a IP PBX for small company. </p>
<h3>Case studies:</h3>
<ul>
<li>
<div align="left">
<a rel="nofollow" target="_blank" href="http://www.2n.cz/news/press_releases/case_study_small_office.html?">
<strong>Small office</strong></a></div>
Voice Blue Enterprise is a complete IP solution for
companies in no PSTN line Areas.<br>
<br>
</li>
<li>
<div align="left">
<a rel="nofollow" target="_blank" href="http://www.2n.cz/news/press_releases/case_study_large_company.html?">
<strong>Large company</strong></a></div>
VoiceBlue Enterprise provides control of outgoing call
routing, aback-up Internet connection option and many more
advanced functions.<br>
</li>
<li>
<div align="left">
<a rel="nofollow" target="_blank" href="http://www.2n.cz/news/press_releases/case_study_voiceblue_enterprise.html?">
<strong>Typical use of VoiceBlue Enterprise for a
company with employees traveling abroad</strong></a><strong>
</strong></div>
By installing VoiceBlue Enterprise GSM gateway in your head
office you relieve your employees on home or foreign
business trips of the necessity to think of their telephone
costs.<br>
</li>
<li>
<div align="left">
<a rel="nofollow" target="_blank" href="http://www.2n.cz/news/press_releases/vbe_with_sipproxy.html?">
<strong>How to use VoiceBlue Enterprise with an external
SIP proxy</strong></a></div>
Containing a SIP proxy server, the Voice Blue Enterprise
VoIP GSM Gateway can substitute a SIP based IP PBX. It can
also be used with any SIP based VoIP PBX.<br>
</li>
<li>
<div align="left">
<a rel="nofollow" target="_blank" href="http://www.2n.cz/news/press_releases/vbe_sippbx.html?">
<strong>Using VoiceBlue Enterprise as a Simple VoIP SIP
PBX</strong></a></div>
This guide will show you how to easily configure the
VoiceBlue Enterprise VoIP GSM Gateway as a SIP proxy for
routing of outgoing calls to GSM.<br>
</li>
<li><strong>
<a rel="nofollow" target="_blank" href="http://www.2n.cz/news/press_releases/cisco-call-manager-express-voip-gsm-gateways.html?">
Configuring Cisco Call Manager Express with VoiceBlue VoIP
GSM Gateways</a></strong></li>
</ul>
<h4 align="left">Features:<img src="http://www.2n.cz/images2/obrazek/172/onpage.jpg" align="right"></h4>
<ul>
<li>
<div>
SIP/H.323 </div>
</li>
<li>
<div>
SIP proxy</div>
</li>
<li>
<div>
Up to 4 GSM channels </div>
</li>
<li>Standard SIP/H.323 clients embedded in one hardware unit
</li>
<li>LCR according to free minutes and GSM providers</li>
<li>SIP proxy Registrar for IP phones included </li>
<li>Intelligent incoming call routing </li>
<li>DISA for incoming calls from GSM </li>
<li>Top voice quality (EFR super sound) </li>
<li>Easy way from Ethernet to Internet over GPRS with "dial
on demand" function</li>
<li>SMS sending and receiving (WEB interface and SMTP/POP3
protocols)</li>
<li>Simple web based configuration </li>
<li>CDR buffer for up to 500,000 records</li>
<li>LOG and statistics saving </li>
<li>Integrated antenna splitter</li>
<li>Worldwide use (all frequencies are supported)</li>
</ul>
<h2><br>
What is Asterisk?</h2>
<p>Asterisk is a software PBX originally developed by Mark
Spencer. This software PBX is designed as an open source, which
means that it can be downloaded from the Internet, namely from:
<a rel="nofollow" target="_blank" href="http://www.asterisk.org/index.php?menu=download">
http://www.asterisk.org/index.php?menu=download</a>,installed
and used without any limitations. The fact that the Asteriskis a
software PBX indicates that VoIP is the native environment for
this PBX. Asterisk supports VoIP using three protocols, which
means that its users can use a wide range of VoIP telephones,
both virtual software type (Skype, Vonage) and hard phones. <br>
<br>
Together with the above mentioned VoIPservices, Asterisk
supports connection of TDM equipment by means of PCI cards
supporting both digital (ISDN PRI/BRI) and analog trunks. Since
the PCI bus on the PC is not unlimited, Asterisk users employ as
much VoIP equipment as possible in order to save the TDM
interface for other services, such as PSTN connection, or use of
analog telephones. This is why most Asterisk users use the
VoiceBlue Lite VoIP GSM Gateway for their GSM calls. You find
below how to interconnect these two types of equipment.<br>
For more information on the Asterisk software solution see
<a rel="nofollow" target="_blank" href="http://www.asterisk.org/">
www.asterisk.org</a>.</p>
<h2>Main scenario</h2>
<p>Suppose we have an IP network to which an Asterisk IP PBX,
several SIP telephones and an VoiceBlue Lite VoIP-GSM gateway
are connected. This typical configuration is shown in the figure
below. Furthermore, suppose that the network is addressed as
shown in the figure and GSM numbers are all numbers starting
with 6,7,8 and containing 9 digits. For configuration
simplicity, use SIM cards from one GSM provider.<br>
Now say that all incoming calls are answered by the gateway,
which replays the invitation message and waits for 10s for
another DTMF dialing. After this timeout, the gateway dials
extension 111, which is a dial-into the operator.<br>
<br>
<img src="http://www.2n.cz/images2/obrazek_click/166/thumbnail.jpg" alt="2N VoiceBlue Lite VoIP GSM Gateway - Main scenario" class="hand" align="middle" border="0"></p>
<h2>VoiceBlue Lite GSM Gateway Configuration</h2>
<p>Step-by-step configuration...ready in 5 minutes:-)</p>
<h4>System Parameters </h4>
<p>Set information on the IP interface only ...address, mask,
default gateway</p>
<p> </p>
<p>
<img src="http://www.2n.cz/images2/obrazek_click/714/thumbnail.jpg" alt="2N VoiceBlue Lite VoIP GSM Gateway - System Parameters" class="hand" align="bottom" border="0"><br>
</p>
<h4>Ethernet Parameters</h4>
<ul>
<li>
<div>
SIP proxy (GSM->IP) </div>
<ul>
<li>instructs the gateway where to send the invite
packet during a GSM-VoIP incoming call ...the Asterisk
address in our case</li>
</ul>
</li>
<li>SIP proxy (IP->GSM)<ul>
<li>instructs the gateway from which IP address the
gateway may receive SIP packets ...the Asterisk address
in our case</li>
</ul>
</li>
<li>SIP registrar<ul>
<li>To enable incoming calls to Asterisk for some units,
register the selected units as Friend types in the
Asterisk system. Set the following parameters in Voice
Blue Lite for registration:<ul>
<li>SIP registrar ...equipment for which the gateway
is to be registered (Asterisk)</li>
<li>Username ...username under which the gateway
shall be registered</li>
<li>Password ...registration password</li>
</ul>
</li>
</ul>
</li>
</ul>
<p>
<img src="http://www.2n.cz/images2/obrazek_click/715/thumbnail.jpg" alt="2N VoiceBlue Lite VoIP GSM Gateway - Ethernet Parameters" class="hand" align="bottom" border="0"><br>
</p>
<h4>Assignment to Groups</h4>
<p><br>
SinceSIM cards from one GSM provider are used, all you have to
do is makesure that all GSM modules have been assigned to the
first group.</p>
<p> </p>
<p>
<img src="http://www.2n.cz/images2/obrazek_click/716/thumbnail.jpg" alt="2N VoiceBlue Lite VoIP GSM Gateway - Assignment to Groups" class="hand" align="bottom" border="0"><br>
</p>
<h4>Network List</h4>
<p>Create a network list containing the prefix of your GSM
destinations...remember to keep the default number of digits (9
in our case).<br>
</p>
<p>
<img src="http://www.2n.cz/images2/obrazek_click/717/thumbnail.jpg?nocache" alt="2N VoiceBlue Lite VoIP GSM Gateway - Network list" class="hand" align="bottom" border="0"><br>
</p>
<h4>LCR Table</h4>
<p>Finally,sum up all settings in a single LCR table defining
that all callednumbers that match the network list 1 shall be
routed via GSM group 1.</p>
<p> </p>
<p>
<img src="http://www.2n.cz/images2/obrazek_click/718/thumbnail.jpg?nocache" alt="2N VoiceBlue Lite VoIP GSM Gateway - LCR Table" class="hand" align="bottom" border="0"><br>
</p>
<h4>Incoming Calls</h4>
<p><br>
As far as incoming calls are concerned, the Voice Blue Lite GSM
Gateway has a relatively wide choice of possibilities. Besides
call ignoring and rejection, you can mainly receive incoming
calls as follows:</p>
<ul>
<li>
<div>
Receive a call and replay the DISA message</div>
<ul>
<li>
<div>
A simply modifiable message is stored in the gateway
and replayed whenever an incoming call is answered.
During this message, the calling subscriber can dial
in to the required extension. </div>
</li>
</ul>
</li>
<li>
<div>
Receive a call with the second dial tone</div>
<ul>
<li>
<div>
The same as with DISA; the only difference is that
the user hears the second dial tone instead of the
message.</div>
</li>
</ul>
</li>
<li>
<div>
Receive a call and forward it to the operator
immediately</div>
<ul>
<li>
<div>
An incoming call can be forwarded to the operator
either immediately or after a timeout.</div>
</li>
</ul>
</li>
</ul>
<p>The following configuration can be used as an example:<br>
</p>
<p>
<img src="http://www.2n.cz/images2/obrazek_click/719/thumbnail.jpg?nocache" alt="2N VoiceBlue Lite VoIP GSM Gateway - Incoming calls" class="hand" align="bottom" border="0"><br>
</p>
<h3>Asterisk IP PBX Configuration</h3>
<p>Now add a few lines in the IP PBX configuration for both
proper routing of outgoing calls to the VoiceBlue Lite GSM
Gateway and receiving calls coming from the GSM gateway to
Asterisk.</p>
<h4>Outgoing Calls</h4>
<p>The core of Asterisk connection lies in the /etc/asterisk/extensions.conf
file.</p>
<p>Open this file in your favorite editor and add the following
lines:<br>
exten => _6XXXXXXXX,1,Dial(SIP/${EXTEN:0}@10.0.0.20,,r)<br>
exten => _7XXXXXXXX,1,Dial(SIP/${EXTEN:0}@10.0.0.20,,r)<br>
exten => _8XXXXXXXX,1,Dial(SIP/${EXTEN:0}@10.0.0.20,,r)</p>
<p>Onceyou have saved and closed the file, restart Asterisk and
from now on all calls starting with 6,7,8 should be routed to
the VoiceBlue Lite GSM Gateway. </p>
<h4>Incoming Calls</h4>
<p>It is recommended to make a little restriction for incoming
calls to prevent unauthorized persons from calling over your
system. <br>
Since VoiceBlue Lite works with the SIP, modify the
/etc/asterisk/sip.conf file where the Voice Blue Lite section
could look as follows, for example:</p>
<p>[voice blue]<br>
type=peer<br>
insecure=very<br>
disallow=all<br>
allow=all allow<br>
host=10.0.0.20<br>
username=voice blue<br>
permit=10.0.0.20/255.255.255.255<br>
qualify=yes<br>
can re-invite=no<br>
call-limit=4<br>
</p>
<p>Again, restart Asterisk after saving the file. After that,
Asterisk will be ready to receive calls coming from the
VoiceBlue Lite GSM Gateway.</p>
<h3>What to do in case of troubles?</h3>
<p>The first thing you should do when you find that there is
something wrong is to run the trace in VoiceBlue Lite... this
helps you locate the problem. First of all you can see
immediately whether any SIP messages come to the GSM gateway and
if so, check the called numbers for proper format. If this is OK
too, review the complete communication listing both on the VoIP
and GSM sides. </p>
To start tracing click on Tracing in the CTRL section. For
recommended trace parameters see the figure below.<br>
<br>
<br>
<img src="http://www.2n.cz/images2/obrazek_click/720/thumbnail.jpg?nocache" class="hand" align="bottom" border="0"><h3>
Order numbers: VOICE BLUE ENTERPRISE</h3>
<table class="prodno" cellpadding="3" cellspacing="0" id="table2">
<tr>
<td>VoiceBlueEnterprise � 2 GSM channels with 2 Siemens
MC39i modules 900/1800 MHz (license needed for GPRS, SIP
proxy, SMS, H.323<>SIP translator)505202E </td>
<td class="no">505202E </td>
</tr>
<tr>
<td>VoiceBlueEnterprise � 4 GSM channels with 4 Siemens
MC39i modules 900/1800 MHz (license needed for GPRS, SIP
proxy, SMS, H.323<>SIP translator)</td>
<td class="no">505204E</td>
</tr>
<tr>
<td>VoiceBlueEnterprise - 2 GSM channels with 2 Siemens
MC55 modules 900/1800/1900MHz (license needed for GPRS,
SIP proxy, SMS, H.323<>SIPtranslator)</td>
<td class="no">505212E</td>
</tr>
<tr>
<td>VoiceBlueEnterprise - 4 GSM channels with 2 Siemens
MC55 modules 900/1800/1900MHz (license needed for GPRS,
SIP proxy, SMS, H.323<>SIPtranslator)</td>
<td class="no">505214E</td>
</tr>
<tr>
<td>VoiceBlueEnterprise - 4 GSM channels with Siemens
MC56 modules 850/1800/1900 MHz( license needed for GPRS,
SIP proxy, SMS, H.323<>SIP translator)</td>
<td class="no">505224US</td>
</tr>
<tr>
<td>VoiceBlueEnterprise - 2 GSM channels with Siemens
MC56 modules 850/1800/1900 MHz (license needed for GPRS,
SIP proxy, SMS, H.323<>SIP translator)</td>
<td class="no">505222US</td>
</tr>
<tr>
<td>VoiceBlueEnterprise - 2 GSM channels with 2 Siemens
MC55 modules 900/1800/1900MHz, GB plug (license needed
for GPRS, SIP proxy, SMS, H.323<>SIPtranslator)</td>
<td class="no">505212GB</td>
</tr>
<tr>
<td>VoiceBlueEnterprise - 4 GSM channels with 4 Siemens
MC55 modules 900/1800/1900MHz, GB plug (license needed
for GPRS, SIP proxy, SMS, H.323<>SIPtranslator)</td>
<td class="no">505214GB</td>
</tr>
<tr>
<td>VoiceBlue Enterprise Internet gateway � GPRS &
embedded router</td>
<td class="no">505400E</td>
</tr>
<tr>
<td>VoiceBlue Enterprise SIP proxy+Mobility Extension 10
USR � number of SIP proxy server users</td>
<td class="no">505401E</td>
</tr>
<tr>
<td>VoiceBlue Enterprise SMS server USR � number of SMS
server users</td>
<td class="no">505402E</td>
</tr>
<tr>
<td>VoiceBlue Enterprise SIP+H.323 translator</td>
<td class="no">505403E</td>
</tr>
</table>
<h3>Order numbers: VOICE BLUE LITE</h3>
<table class="prodno" cellpadding="3" cellspacing="0" id="table3">
<tr>
<td>2 channel VoIP GSM gateway (900/1800 MHz)</td>
<td class="no">505002E</td>
</tr>
<tr>
<td>4 channel VoIP GSM gateway (900/1800 MHz)</td>
<td class="no">505004E</td>
</tr>
<tr>
<td>2 channel VoIP GSM gateway (900/1800/1900 MHz)</td>
<td class="no">505032E</td>
</tr>
<tr>
<td>4 channel VoIP GSM gateway (900/1800/1900 MHz)</td>
<td class="no">505034E</td>
</tr>
<tr>
<td>2 channel Wismo Q24Clasic VoIP GSM gateway (900/1800
MHz)</td>
<td class="no">505052E</td>
</tr>
<tr>
<td>2 channel VoIP GSM gateway (850/1800/1900 MHz) with
US power supply</td>
<td class="no">505042US</td>
</tr>
<tr>
<td>4 channel VoIP GSM gateway (850/1800/1900 MHz) with
US power supply</td>
<td class="no">505044US</td>
</tr>
<tr>
<td>4 channel Wismo Q24Clasic VoIP GSM gateway (900/1800
MHz)</td>
<td class="no">505054E</td>
</tr>
<tr>
<td>SMS user licence</td>
<td class="no">502999E</td>
</tr>
<tr>
<td>Mobility Extension licence for 8 users</td>
<td class="no">5050995E</td>
</tr>
<tr>
<td>Mobility Extension licence for 32 users</td>
<td class="no">5050998E</td>
</tr>
<tr>
<td>Mobility Extension licence for 16 users</td>
<td class="no">5050996E</td>
</tr>
<tr>
<td>Mobility Extension licence for 24 users</td>
<td class="no">5050997E</td>
</tr>
<tr>
<td>2 channel Wismo Q24Clasic VoIP GSM gateway (850/1900
MHz)</td>
<td class="no">505052US</td>
</tr>
<tr>
<td>4 channel Wismo Q24Clasic VoIP GSM gateway (850/1900
MHz)</td>
<td class="no">505054US</td>
</tr>
</table>
<hr>
<p align="center"><b>
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<font size="5">CONTACT US</font></a></b></p>
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