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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'>First Issue to be addressed is how many simultaneous
calls and bandwidth availability.<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'>Number of “lines” (numbers) is
not a limitation in it self unless they are all in use.<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>
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color=navy face=Arial><span style='font-size:10.0pt;font-family:Arial;
color:navy'>Cary</span></font></st1:City></st1:place><font size=2 color=navy
face=Arial><span style='font-size:10.0pt;font-family:Arial;color:navy'> Fitch<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>
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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>Christian Victor<br>
<b><span style='font-weight:bold'>Sent:</span></b> Tuesday, March 24, 2009
10:10 AM<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> [asterisk-users] SIP
trunk with > 250 lines</span></font><o:p></o:p></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>Hi!<br>
<br>
A customer of mine wants to connect an asterisk system with 240 to 480 lines to
a PSTN switch. To save the costs for E1 cards and the corresponding E1
mainlines he wants to connect the system to the switch by a SIP trunk.<br>
<br>
Phones will be connected to the server through the same SIP trunk as this will
be some kind of a "hosted pbx".<br>
<br>
Given he finds a provider wich has this much SIP capacity and IP bandwith and
no codec conversion is needed - do you think this is possible with pure
asterisk on a decent system? Is there anything I shoudl watch out for?<br>
<br>
Your help is much appreciated!<br>
<br>
Chris<o:p></o:p></span></font></p>
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