Can you provide the link to the website.<br>
I"ll like to see the specs in more details and the whole product range.<br>
Thanks<br><br><div><span class="gmail_quote">On 5/10/06, <b class="gmail_sendername">Hao Xu</b> <<a href="mailto:hao.xu.cn@gmail.com">hao.xu.cn@gmail.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
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<div><font size="2">
</font><p style="margin: 0cm 0cm 0pt; text-align: left;" align="left"><span style="font-size: 10pt; font-family: Tahoma;" lang="EN-US"><font size="2">Hi<span>, every one</span></font></span></p>
<p style="margin: 0cm 0cm 0pt; text-align: left;" align="left"><span style="font-size: 10pt; font-family: Tahoma;" lang="EN-US"><span></span></span><font size="2"> </font></p>
<p style="margin: 0cm 0cm 0pt; text-align: left;" align="left"><span style="font-size: 10pt; font-family: Tahoma;" lang="EN-US"><span><font size="2">I'd like to introduce some new feature of our
products.</font></span></span></p>
<p style="margin: 0cm 0cm 0pt; text-align: left;" align="left"><span style="font-size: 10pt; font-family: Tahoma;" lang="EN-US"><span></span></span><font size="2"> </font></p>
<p style="margin: 0cm 0cm 0pt; text-align: left;" align="left"><font size="2"><b><span style="font-size: 10pt; font-family: Tahoma;" lang="EN-US">mg3000-r
</span></b><b><span style="font-size: 10pt; font-family: Arial;" lang="EN-US">fxo gateway
</span></b><b><span style="font-size: 10pt; font-family: Tahoma;" lang="EN-US">provides
more feature to work with asterisk.</span></b></font></p>
<p style="margin: 0cm 0cm 0pt; text-align: left;" align="left"><span style="font-size: 10pt; font-family: Tahoma;" lang="EN-US"><font size="2"> </font></span></p>
<p style="margin: 0cm 0cm 0pt; text-align: left;" align="left"><span style="font-size: 10pt; font-family: Tahoma;" lang="EN-US"><font size="2">1.play
asterisk ivr with no interuption.</font></span></p>
<p style="margin: 0cm 0cm 0pt; text-align: left;" align="left"><span style="font-size: 10pt; font-family: Tahoma;" lang="EN-US"><font size="2">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</font></span><span style="font-size: 10pt; font-family: Arial;" lang="EN-US"><font size="2">
feel</font></span><span style="font-size: 10pt; font-family: Tahoma;" lang="EN-US"><font size="2">
friendly.<span> </span></font></span></p>
<p style="margin: 0cm 0cm 0pt; text-align: left;" align="left"><span style="font-size: 10pt; font-family: Tahoma;" lang="EN-US"><font size="2">2. pbx
voip/pstn inteleged route.</font></span></p>
<p style="margin: 0cm 0cm 0pt; text-align: left;" align="left"><font size="2"><span style="font-size: 10pt; font-family: Tahoma;" lang="EN-US">when you
make pbx connect to voip/asterisk,</span><span style="font-size: 10pt; font-family: Arial;" lang="EN-US"> 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 style="font-size: 10pt; font-family: Tahoma;" lang="EN-US"></span></font></p>
<p style="margin: 0cm 0cm 0pt; text-align: left;" align="left"><span style="font-size: 10pt; font-family: Tahoma;" lang="EN-US"><font size="2">3. pstn
caller number transfer.</font></span></p>
<p style="margin: 0cm 0cm 0pt; text-align: left;" align="left"><span style="font-size: 10pt; font-family: Arial;" lang="EN-US"><font size="2">When pstn
call in, the mg3000-R start voip call to the asterisk using pstn caller number
instead of gateway number.</font></span><span style="font-size: 10pt; font-family: Tahoma;" lang="EN-US"></span></p>
<p style="margin: 0cm 0cm 0pt; text-align: left;" align="left"><span style="font-size: 10pt; font-family: Tahoma;" lang="EN-US"><font size="2">4. multy
region pstn singal support. </font></span></p>
<p style="margin: 0cm 0cm 0pt; text-align: left;" align="left"><span style="font-size: 10pt; font-family: Arial;" lang="EN-US"><font size="2">By using
MindSpeed technology, it integrated many region's pstn singal.</font></span><span style="font-size: 10pt; font-family: Tahoma;" lang="EN-US"></span></p>
<p style="margin: 0cm 0cm 0pt;"><span lang="EN-US"><font size="2"><font size="3"> </font></font></span></p>
<p style="margin: 0cm 0cm 0pt;"><span lang="EN-US"><font size="2"><font size="3"> </font></font></span></p>
<p style="margin: 0cm 0cm 0pt;"><font size="2"><b><span lang="EN-US"><font size="3">Why we choose
a voip fxo gateway while not a asterisk card.</font></span></b></font></p>
<p style="margin: 0cm 0cm 0pt;"><font size="2"><b><span lang="EN-US"><font size="3"> </font></span></b></font></p>
<p style="margin: 0cm 0cm 0pt 18pt; text-indent: -18pt;"><span lang="EN-US"><span><font size="2"><font size="3">1.</font><span>
</span></font></span></span><span lang="EN-US"><font size="2"><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> </span>we sugguest you to use voip gateway on
more than 4 phone line system.</font></font></span></p>
<p style="margin: 0cm 0cm 0pt 18pt; text-indent: -18pt;"><span lang="EN-US"><span><font size="2"><font size="3">2.</font><span>
</span></font></span></span><span lang="EN-US"><font size="2"><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></font></span></p>
<p style="margin: 0cm 0cm 0pt 18pt; text-indent: -18pt;"><span lang="EN-US"><span><font size="2"><font size="3">3.</font><span>
</span></font></span></span><span lang="EN-US"><font size="2"><font size="3">There are many analog voip
gateway producer, the price is cheap, especially for MG3000-R
fxs.</font></font></span></p>
<p style="margin: 0cm 0cm 0pt;"><span lang="EN-US"><font size="2"><font size="3"> </font></font></span></p>
<p style="margin: 0cm 0cm 0pt;"><font size="2"><b><span lang="EN-US"><font size="3">How MG3000-R
work with Asterisk feature.</font></span></b></font></p>
<p style="margin: 0cm 0cm 0pt;"><span lang="EN-US"><font size="2"><font size="3"> </font></font></span></p>
<p style="margin: 0cm 0cm 0pt;"><span lang="EN-US"><font size="2"><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></font></span></p>
<p style="margin: 0cm 0cm 0pt;"><span lang="EN-US"><font size="2"><font size="3">If<span> </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></font></span></p>
<p style="margin: 0cm 0cm 0pt;"><span lang="EN-US"><font size="2"><font size="3"> </font></font></span></p>
<p style="margin: 0cm 0cm 0pt;"><span lang="EN-US"><font size="2"><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></font></span></p>
<p style="margin: 0cm 0cm 0pt;"><span lang="EN-US"><font size="2"><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></font></span></p>
<p style="margin: 0cm 0cm 0pt;"><span lang="EN-US"><font size="2"><font size="3"> </font></font></span></p>
<p style="margin: 0cm 0cm 0pt;"><font size="2"><b><span lang="EN-US"><font size="3">How MG3000-R
backup voip with the PSTN line </font></span></b></font></p>
<p style="margin: 0cm 0cm 0pt;"><span lang="EN-US"><font size="2"><font size="3"> </font></font></span></p>
<p style="margin: 0cm 0cm 0pt;"><span lang="EN-US"><font size="2"><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></font></span></p>
<p style="margin: 0cm 0cm 0pt;"><span lang="EN-US"><font size="2"><font size="3"> </font></font></span></p>
<p style="margin: 0cm 0cm 0pt;"><span lang="EN-US"><font size="2"><font size="3">The
voip change to pstn condition is: when voip service is unavailable or the power
is loose.</font></font></span></p>
<p style="margin: 0cm 0cm 0pt;"><span lang="EN-US"><font size="2"><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></font></span></p>
<p style="margin: 0cm 0cm 0pt;"><span lang="EN-US"><font size="2"><font size="3"> </font></font></span></p>
<p style="margin: 0cm 0cm 0pt;"><span lang="EN-US"><font size="2"><font size="3">For
more info please<span> contact us <a href="mailto:hao.xu.cn@@gmail.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">hao.xu.cn@@gmail.com</a></span></font></font></span></p>
<p style="margin: 0cm 0cm 0pt;"><span lang="EN-US"><font size="2"><font size="3"><span></span></font></font></span><font size="2"> </font></p></div></div>
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