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<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Signate sells a single server that can get
you to the call volumes you need. <o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Paul Mahler<o:p></o:p></span></font></p>
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10.0pt;font-family:Arial;color:navy'><a href="mailto:pmahler@signate.com">pmahler@signate.com</a><o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>www.signate.com<o:p></o:p></span></font></p>
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10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
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<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=2
face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b><span style='font-weight:
bold'>On Behalf Of </span></b>Vic<br>
<b><span style='font-weight:bold'>Sent:</span></b> Saturday, January 28, 2006
7:16 PM<br>
<b><span style='font-weight:bold'>To:</span></b>
asterisk-users@lists.digium.com<br>
<b><span style='font-weight:bold'>Subject:</span></b> Re: [Asterisk-Users]
5,000 concurrent calls system rollout question</span></font><o:p></o:p></p>
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<p class=MsoNormal><font size=3 face="MS PGothic"><span style='font-size:12.0pt'><o:p> </o:p></span></font></p>
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<p><font size=3 face="MS PGothic"><span style='font-size:12.0pt'>Hi, Zoa, <o:p></o:p></span></font></p>
<p><font size=3 face="MS PGothic"><span style='font-size:12.0pt'>yes, these
calls are from SIP to SIP. We will have more than 3000 (more like
5000)concurrent calls come into system and we will need to handle them. <o:p></o:p></span></font></p>
<p><font size=3 face="MS PGothic"><span style='font-size:12.0pt'>We will also
need an IVR function as well. <o:p></o:p></span></font></p>
<p><font size=3 face="MS PGothic"><span style='font-size:12.0pt'>I am not up
to speed on Asterisk yet, so, I am a little bit confused by all the different
ways of doing it. Someone is talking about IAX:</span></font><font
face="Times New Roman"><span style='font-family:"Times New Roman"'> </span></font>
I think it can only be used between Asterisk servers, right? <o:p></o:p></p>
<p><font size=3 face="MS PGothic"><span style='font-size:12.0pt'>In this particula
rscenario we are getting calls as SIP directly from carrier, so we will not
need to do any conversion (I think). We just route the calls to the
destination, that's it. <o:p></o:p></span></font></p>
<p><font size=3 face="MS PGothic"><span style='font-size:12.0pt'>Any
suggestions on how to proceed? Can Asterisk do it? <o:p></o:p></span></font></p>
<p><font size=3 face="MS PGothic"><span style='font-size:12.0pt'>I read
somewhere that it takes about 30 MHz per one voice channel, so if we want to
have 5,000 calls, we will need 150,000 MHz? Thats like 50 3 GHz machines...
Not going to fly with our people.</span></font><font face="Times New Roman"><span
style='font-family:"Times New Roman"'> </span></font> <o:p></o:p></p>
<p><font size=3 face="MS PGothic"><span style='font-size:12.0pt'>Or do 30 MHz
are only necessary for transcoding? In other words, if it comes in as SIP and
we keep it that way, can</span></font><font face="Times New Roman"><span
style='font-family:"Times New Roman"'> </span></font>we make it<font
face="Times New Roman"><span style='font-family:"Times New Roman"'> </span></font>a
bt more feasible number?<font face="Times New Roman"><span style='font-family:
"Times New Roman"'> </span></font> <o:p></o:p></p>
<p><font size=3 face="Times New Roman"><span style='font-size:12.0pt;
font-family:"Times New Roman"'> </span></font> <o:p></o:p></p>
<p><font size=3 face="Times New Roman"><span style='font-size:12.0pt;
font-family:"Times New Roman"'> </span></font><b><i><span
style='font-weight:bold;font-style:italic'>Zoa <zoachien@securax.org></span></i></b>
wrote: <o:p></o:p></p>
<blockquote style='border:none;border-left:solid #1010FF 1.5pt;padding:0in 0in 0in 4.0pt;
margin-left:3.75pt;margin-top:5.0pt;margin-bottom:5.0pt'>
<p class=MsoNormal><font size=3 face="MS PGothic"><span style='font-size:
12.0pt'><br>
It can be done, are those 3000 calls sip to sip ? If so it could easily<br>
be done, if they are not sip to sip you will need a bunch of servers.<br>
<br>
Zoa.<br>
<br>
Vic wrote:<br>
<br>
> Hi,<br>
><br>
> we are currently considering different options for rolling out a large<br>
> scale IP PBX to handle around 3,000 + concurrent calls.<br>
><br>
> Can this be done with Asterisk? Has it been done before?<br>
><br>
> I really would like an input on this.<br>
><br>
> Thanks!<br>
><br>
>------------------------------------------------------------------------<br>
><br>
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