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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Hello,</span></font></p>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> </span></font></p>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>I need help in creating a simple PSTN Gateway. This is the scenario:</span></font></p>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> </span></font></p>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>-I have one client sending me VoIP traffic (they don’t have
asterisk, so IAX is out of the picture for me) and I need to validate that
traffic (only accept calls coming from his IP). After that I would terminate
the calls to the PSTN network and keep logs for billing purposes.</span></font></p>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> </span></font></p>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>-I have a TE405P board and only one T1 worth of phone lines (24)
connected to it using an Adtran TA750 channel bank.</span></font></p>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> </span></font></p>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>Is Asterisk capable of handling multiple incoming VoIP calls arriving
from the same source (IP) or do I need to get something else to take the
incoming traffic and pass it on to Asterisk? (I’ve read about using SER
as a SIP proxy, but it’s not clear to me wheather I need it or not). Can
I use the OpenH.323 module to take care of the incoming VoIP traffic?</span></font></p>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> </span></font></p>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>I am totally new to all this. Any help will be really appreciated.</span></font></p>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> </span></font></p>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>Please give me your input on which is the best implementation of a
VoIP->PSTN gateway and the some sample configuration files for the
plattforms involved (Asterisk, SER, OpenH323, etc.)</span></font></p>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> </span></font></p>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>Thanks in advance.</span></font></p>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> </span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'> </span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Alejandro Sosa.</span></font></p>
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font-family:Arial'> </span></font></p>
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