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<DIV><FONT color=#0000ff face=Arial size=2><SPAN class=483223618-06082004>Will
you have E1s? will you restrict users to 729 or will you allow other codecs?
will most calls be from SIP to SIP? or SIP to E1 lines?</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN
class=483223618-06082004></SPAN></FONT> </DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN
class=483223618-06082004>MATT---</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN
class=483223618-06082004></SPAN></FONT> </DIV>
<BLOCKQUOTE style="MARGIN-RIGHT: 0px">
<DIV align=left class=OutlookMessageHeader dir=ltr><FONT face=Tahoma
size=2>-----Original Message-----<BR><B>From:</B> Sebastian Nocetti
[mailto:sebastian@interband.com.ar]<BR><B>Sent:</B> Friday, August 06, 2004
12:53 PM<BR><B>To:</B> asterisk-users@lists.digium.com<BR><B>Subject:</B>
[Asterisk-Users] ASTERISK AND 120 CONCURRENT CALLS<BR><BR></DIV></FONT>
<DIV><SPAN class=846005216-06082004><FONT face=Arial size=2>hello all, does
anyone has experiencie using asterisk with a digium CARD using G729 managing
120 concurrent calls with SIP and/or H323??? I wanna know if Asterisk is
stable doing this....because we wanna implement it in some
locations!!</FONT></SPAN></DIV>
<DIV><SPAN class=846005216-06082004><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=846005216-06082004><FONT face=Arial size=2>Thanks
All!!</FONT></SPAN></DIV>
<DIV><SPAN class=846005216-06082004><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=846005216-06082004><FONT face=Arial
size=2>Sebastian.</FONT></SPAN></DIV></BLOCKQUOTE></BODY></HTML>