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<DIV><FONT face=Arial size=2><SPAN class=016131803-09102003>I have researched
the topic and would be interested to see the "theory versus practice" of using
Asterisk versus Asterisk with SER to handle SIP to SIP and SIP to PSTN
calls.</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
class=016131803-09102003>Config:</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN class=016131803-09102003>1) SIP ----
NAT ----- Internet ------ SER ----- Internet ----- NAT ----
SIP (i.e. no Asterisk in the picture for SIP to SIP
calls)</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN class=016131803-09102003>2) PSTN (phone call)
---- Asterisk (via SIP) ------ SER (i.e. any time traffic is
to be sent/received to/from PSTN, use Asterisk as GW)</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
class=016131803-09102003></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN class=016131803-09102003>Both SER and
ASTERISK are in the same LAN.</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN class=016131803-09102003>Anybody with
experience in this type of set up willing to share war
stories?</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN class=016131803-09102003>Would this
configuration help those members of this list complaining Asterisk is dropping
registered Budgetone/SIP phones?</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
class=016131803-09102003></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN
class=016131803-09102003>Regards,</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
class=016131803-09102003>Uriel</SPAN></FONT></DIV></BODY></HTML>