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<P class=MsoNormal
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align=left><SPAN lang=EN-US
style="FONT-SIZE: 10pt; FONT-FAMILY: Tahoma; mso-font-kerning: 0pt">Hi<SPAN
class=003204008-10052006>, every one</SPAN></SPAN></P>
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class=003204008-10052006></SPAN></SPAN> </P>
<P class=MsoNormal
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align=left><SPAN lang=EN-US
style="FONT-SIZE: 10pt; FONT-FAMILY: Tahoma; mso-font-kerning: 0pt"><SPAN
class=003204008-10052006>I'd like to introduce some new feature of our
products.</SPAN></SPAN></P>
<P class=MsoNormal
style="MARGIN: 0cm 0cm 0pt; TEXT-ALIGN: left; mso-layout-grid-align: none"
align=left><SPAN lang=EN-US
style="FONT-SIZE: 10pt; FONT-FAMILY: Tahoma; mso-font-kerning: 0pt"><SPAN
class=003204008-10052006></SPAN></SPAN> </P>
<P class=MsoNormal
style="MARGIN: 0cm 0cm 0pt; TEXT-ALIGN: left; mso-layout-grid-align: none"
align=left><B style="mso-bidi-font-weight: normal"><SPAN lang=EN-US
style="FONT-SIZE: 10pt; FONT-FAMILY: Tahoma; mso-font-kerning: 0pt">mg3000-r
</SPAN></B><B style="mso-bidi-font-weight: normal"><SPAN lang=EN-US
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial; mso-font-kerning: 0pt">fxo gateway
</SPAN></B><B style="mso-bidi-font-weight: normal"><SPAN lang=EN-US
style="FONT-SIZE: 10pt; FONT-FAMILY: Tahoma; mso-font-kerning: 0pt">provides
more feature to work with asterisk.<?xml:namespace prefix = o ns =
"urn:schemas-microsoft-com:office:office" /><o:p></o:p></SPAN></B></P>
<P class=MsoNormal
style="MARGIN: 0cm 0cm 0pt; TEXT-ALIGN: left; mso-layout-grid-align: none"
align=left><SPAN lang=EN-US
style="FONT-SIZE: 10pt; FONT-FAMILY: Tahoma; mso-font-kerning: 0pt"><o:p> </o:p></SPAN></P>
<P class=MsoNormal
style="MARGIN: 0cm 0cm 0pt; TEXT-ALIGN: left; mso-layout-grid-align: none"
align=left><SPAN lang=EN-US
style="FONT-SIZE: 10pt; FONT-FAMILY: Tahoma; mso-font-kerning: 0pt">1.play
asterisk ivr with no interuption.<o:p></o:p></SPAN></P>
<P class=MsoNormal
style="MARGIN: 0cm 0cm 0pt; TEXT-ALIGN: left; mso-layout-grid-align: none"
align=left><SPAN lang=EN-US
style="FONT-SIZE: 10pt; FONT-FAMILY: Tahoma; mso-font-kerning: 0pt">when the
mg3000-r received call from co line, it wouldn't conect instantly.instead, it
start call to asterisk ivr first,when the ivr ready, it connect the co line.
this feature make user</SPAN><SPAN lang=EN-US
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial; mso-font-kerning: 0pt">
feel</SPAN><SPAN lang=EN-US
style="FONT-SIZE: 10pt; FONT-FAMILY: Tahoma; mso-font-kerning: 0pt">
friendly.<SPAN style="mso-spacerun: yes"> </SPAN><o:p></o:p></SPAN></P>
<P class=MsoNormal
style="MARGIN: 0cm 0cm 0pt; TEXT-ALIGN: left; mso-layout-grid-align: none"
align=left><SPAN lang=EN-US
style="FONT-SIZE: 10pt; FONT-FAMILY: Tahoma; mso-font-kerning: 0pt">2. pbx
voip/pstn inteleged route.<o:p></o:p></SPAN></P>
<P class=MsoNormal
style="MARGIN: 0cm 0cm 0pt; TEXT-ALIGN: left; mso-layout-grid-align: none"
align=left><SPAN lang=EN-US
style="FONT-SIZE: 10pt; FONT-FAMILY: Tahoma; mso-font-kerning: 0pt">when you
make pbx connect to voip/asterisk,</SPAN><SPAN lang=EN-US
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial; mso-font-kerning: 0pt"> how to make
voip more stable. MG3000-R could detect the voip quanlity, when voip line
failed, it change to pstn line automaticly.</SPAN><SPAN lang=EN-US
style="FONT-SIZE: 10pt; FONT-FAMILY: Tahoma; mso-font-kerning: 0pt"><o:p></o:p></SPAN></P>
<P class=MsoNormal
style="MARGIN: 0cm 0cm 0pt; TEXT-ALIGN: left; mso-layout-grid-align: none"
align=left><SPAN lang=EN-US
style="FONT-SIZE: 10pt; FONT-FAMILY: Tahoma; mso-font-kerning: 0pt">3. pstn
caller number transfer.<o:p></o:p></SPAN></P>
<P class=MsoNormal
style="MARGIN: 0cm 0cm 0pt; TEXT-ALIGN: left; mso-layout-grid-align: none"
align=left><SPAN lang=EN-US
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial; mso-font-kerning: 0pt">When pstn
call in, the mg3000-R start voip call to the asterisk using pstn caller number
instead of gateway number.</SPAN><SPAN lang=EN-US
style="FONT-SIZE: 10pt; FONT-FAMILY: Tahoma; mso-font-kerning: 0pt"><o:p></o:p></SPAN></P>
<P class=MsoNormal
style="MARGIN: 0cm 0cm 0pt; TEXT-ALIGN: left; mso-layout-grid-align: none"
align=left><SPAN lang=EN-US
style="FONT-SIZE: 10pt; FONT-FAMILY: Tahoma; mso-font-kerning: 0pt">4. multy
region pstn singal support. <o:p></o:p></SPAN></P>
<P class=MsoNormal
style="MARGIN: 0cm 0cm 0pt; TEXT-ALIGN: left; mso-layout-grid-align: none"
align=left><SPAN lang=EN-US
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial; mso-font-kerning: 0pt">By using
MindSpeed technology, it integrated many region's pstn singal.</SPAN><SPAN
lang=EN-US
style="FONT-SIZE: 10pt; FONT-FAMILY: Tahoma; mso-font-kerning: 0pt"><o:p></o:p></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US><o:p><FONT
size=3> </FONT></o:p></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US><o:p><FONT
size=3> </FONT></o:p></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><B
style="mso-bidi-font-weight: normal"><SPAN lang=EN-US><FONT size=3>Why we choose
a voip fxo gateway while not a asterisk card.<o:p></o:p></FONT></SPAN></B></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><B
style="mso-bidi-font-weight: normal"><SPAN lang=EN-US><o:p><FONT
size=3> </FONT></o:p></SPAN></B></P>
<P class=MsoNormal
style="MARGIN: 0cm 0cm 0pt 18pt; TEXT-INDENT: -18pt; mso-list: l0 level1 lfo1; tab-stops: list 18.0pt"><SPAN
lang=EN-US style="mso-fareast-font-family: 'Times New Roman'"><SPAN
style="mso-list: Ignore"><FONT size=3>1.</FONT><SPAN
style="FONT: 7pt 'Times New Roman'">
</SPAN></SPAN></SPAN><SPAN lang=EN-US><FONT size=3>voice process is a realtime
task. PC operate system is not a realtime one. Voip gateway use its own dsp to
do voice process while asterisk card use PC CPU to do this. Just like the DVD
decode, on heavy task, voip gateway hardware will do better. So<SPAN
style="mso-spacerun: yes"> </SPAN>we sugguest you to use voip gateway on
more than 4 phone line system.</FONT></SPAN></P>
<P class=MsoNormal
style="MARGIN: 0cm 0cm 0pt 18pt; TEXT-INDENT: -18pt; mso-list: l0 level1 lfo1; tab-stops: list 18.0pt"><SPAN
lang=EN-US style="mso-fareast-font-family: 'Times New Roman'"><SPAN
style="mso-list: Ignore"><FONT size=3>2.</FONT><SPAN
style="FONT: 7pt 'Times New Roman'">
</SPAN></SPAN></SPAN><SPAN lang=EN-US><FONT size=3>Asterisk PC+ voip gateway
model, it is easy to expand to over 100 user. In this scale, you can not plug so
many card into one PC.</FONT></SPAN></P>
<P class=MsoNormal
style="MARGIN: 0cm 0cm 0pt 18pt; TEXT-INDENT: -18pt; mso-list: l0 level1 lfo1; tab-stops: list 18.0pt"><SPAN
lang=EN-US style="mso-fareast-font-family: 'Times New Roman'"><SPAN
style="mso-list: Ignore"><FONT size=3>3.</FONT><SPAN
style="FONT: 7pt 'Times New Roman'">
</SPAN></SPAN></SPAN><SPAN lang=EN-US><FONT size=3>There are many analog voip
gateway producer, the price is cheap, especially for MG3000-R
fxs.</FONT></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US><o:p><FONT
size=3> </FONT></o:p></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><B
style="mso-bidi-font-weight: normal"><SPAN lang=EN-US><FONT size=3>How MG3000-R
work with Asterisk feature.<o:p></o:p></FONT></SPAN></B></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US><o:p><FONT
size=3> </FONT></o:p></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US><FONT size=3>For
FXS voip gateway, interoperate with Asterisk is easy. There are two
requirements: one is sip interoperability. The other is DTMF transfer
model.</FONT></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US><FONT
size=3>If<SPAN style="mso-spacerun: yes"> </SPAN>a voip gateway can make
call with asterisk, that could to say sip interoperability is ok. If the auto
attendant service is ok, that is mean DTMF transfer is ok.. The other things
will be no problem. </FONT></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US><o:p><FONT
size=3> </FONT></o:p></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US><FONT
size=3>Howerver there are some limits in Asterisk, especially on transcoding. If
we use g.711, all is ok. but we are normal use g.723 or g.729. when we want to
use conference service. Asterisk need to change codec to
G.711.</FONT></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US><FONT size=3>The
other things you must be careful. When a call setup successfully, there need
call original voip gateway, and also the call termination voip gateway. The
interoperation of such two type gateway is also important.</FONT></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US><o:p><FONT
size=3> </FONT></o:p></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><B
style="mso-bidi-font-weight: normal"><SPAN lang=EN-US><FONT size=3>How MG3000-R
backup voip with the PSTN line <o:p></o:p></FONT></SPAN></B></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US><o:p><FONT
size=3> </FONT></o:p></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US><FONT
size=3>MG3000-R 4s4o owned 4 fxo and 4 fxs ports. When you work in backup mode,
4 fxs ports connect to pbx trunk line, 4 fxo ports connect to CO line. User
start a voip call from fxs port, When a voip call failed, MG3000-R will
automatically dial on the pstn line through fxo port. The end user use the
gateway like it was on voip line. So the voip call quanlity is protected with
pstn backup line.</FONT></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US><o:p><FONT
size=3> </FONT></o:p></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US><FONT size=3>The
voip change to pstn condition is: when voip service is unavailable or the power
is loose.</FONT></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US><FONT size=3>It
is very difference than lifeline. Lifeline is a n to 1 protection. This is 1:1
protection. Lifeline work only when the power is loose, it has no use when the
network is down.</FONT></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US><o:p><FONT
size=3> </FONT></o:p></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US><FONT size=3>For
more info please<SPAN class=003204008-10052006> contact us <A
href="mailto:hao.xu.cn@@gmail.com">hao.xu.cn@@gmail.com</A></SPAN></FONT></SPAN></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><SPAN lang=EN-US><FONT
size=3><SPAN
class=003204008-10052006></SPAN></FONT></SPAN> </P></FONT></DIV></BODY></HTML>