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<DIV><SPAN class=944282016-19062006><FONT face=Arial color=#0000ff
size=2>Remember to add the RTP, UDP and IP overheads.</FONT></SPAN></DIV>
<DIV><SPAN class=944282016-19062006><FONT face=Arial color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=944282016-19062006><FONT face=Arial color=#0000ff size=2>And
then just do the math.</FONT></SPAN></DIV>
<BLOCKQUOTE dir=ltr
style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px solid; MARGIN-RIGHT: 0px">
<DIV class=OutlookMessageHeader dir=ltr align=left><FONT face=Tahoma
size=2>-----Original Message-----<BR><B>From:</B>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com]<B>On Behalf Of </B>William
Piper<BR><B>Sent:</B> 19 June 2006 17:12<BR><B>To:</B> Asterisk Users Mailing
List - Non-Commercial Discussion<BR><B>Subject:</B> Re: [Asterisk-Users] How
to use a data T-1?<BR><BR></FONT></DIV>
<DIV>Depends on the codec. If you are using ulaw, you will only be able to
have about 23 calls. If you use g729 you can have as many as 187 simultanious
calls on a data T1.</DIV>
<DIV> </DIV>
<DIV>Remember, you have 1544Kbs of bandwidth. </DIV>
<DIV>g279=8Kbs per call</DIV>
<DIV>uLaw=64Kbs per call</DIV>
<DIV> </DIV>
<DIV>Just do the math.</DIV>
<DIV> </DIV>
<DIV>bp<BR><BR> </DIV>
<DIV><SPAN class=gmail_quote>On 6/19/06, <B class=gmail_sendername>Warren</B>
<<A href="mailto:warren-lists@icruise.com">warren-lists@icruise.com</A>>
wrote:</SPAN>
<BLOCKQUOTE class=gmail_quote
style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">Steve,<BR><BR>I
want to end up with a system that will let me send and receive
voice<BR>calls. I guess what I want to do depends on the best way
to do that. <BR>Can I do more than 23 (decent sounding) voice calls on a
data T-1 with<BR>someone else handling the final part of the call to the
copper for me?<BR>If so than that is my likely final destination.<BR><BR>I
have a channelized voice T-1 currently plugged into my meridian <BR>system,
but I would like (if realistically possible) to do as much of<BR>this over
IP as possible for maximum flexibility. Is that a pipe
dream<BR>or just silly given the current state of technology?<BR><BR>I am
lucky enough to work for a company that is letting me take my time <BR>with
this, test the various options and come up with the
proper<BR>solution. I am assuming (I know: dumb to assume) at
this point that<BR>VoIP over a T-1 to a provider that can then route it to
hard phones for<BR>me would be the way to go. Similarly, I would
point my 800 number to a <BR>DiD hosted by a VoIP provider that would then
route the call back to<BR>me. If that is an incorrect assumption,
please let me
know.<BR><BR>Regards,<BR>Warren</BLOCKQUOTE></DIV></BLOCKQUOTE></BODY></HTML>