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<DIV><FONT face="Comic Sans MS" size=2>Hi</FONT></DIV>
<DIV><FONT face="Comic Sans MS" size=2>I have ser up and running and i can make
calls from/to SIP UA but i want to test PSTN termination as well</FONT></DIV>
<DIV><FONT face="Comic Sans MS" size=2>for that purpose i have one cisco voice
box at the end</FONT></DIV>
<DIV><FONT face="Comic Sans MS" size=2>we have dial peer plan as 441 and
442</FONT></DIV>
<DIV><FONT face="Comic Sans MS" size=2>now i want to make call to a pstn no say
1126102911</FONT></DIV>
<DIV><FONT face="Comic Sans MS" size=2>how do i do that..? i have tried to
follow the mailing list but all vain..cna somebody help me to test this
out.</FONT></DIV>
<DIV><FONT face="Comic Sans MS" size=2></FONT> </DIV>
<DIV><FONT face="Comic Sans MS" size=2>PSTN relevant ser.cfg</FONT></DIV>
<DIV><FONT face="Comic Sans MS" size=2></FONT> </DIV>
<DIV><FONT face="Comic Sans MS" size=2>if
(uri=~"sip:[4-9]+@(mydomain|202\.71\.135\.219)")
{<BR> t_relay_to_udp("212.74.110.243",
"5060");<BR> break;<BR>};</FONT></DIV>
<DIV><FONT face="Comic Sans MS" size=2></FONT> </DIV>
<DIV><FONT face="Comic Sans MS" size=2></FONT> </DIV>
<DIV><FONT face="Comic Sans MS" size=2>regards,</FONT></DIV>
<DIV><FONT face="Comic Sans MS" size=2>Madan</FONT></DIV></BODY></HTML>