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<DIV><FONT face=Verdana>check rtp.conf</FONT></DIV>
<DIV><FONT face=Verdana></FONT> </DIV>
<DIV><FONT face=Verdana>-Kannaiyan</FONT></FONT></DIV>
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<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B>
<A title=boris.vincent@mindspeed.com
href="mailto:boris.vincent@mindspeed.com">boris.vincent@mindspeed.com</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=asterisk-users@lists.digium.com
href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Monday, September 06, 2004 6:15
PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> [Asterisk-Users] SIP rtp port
forcing</DIV>
<DIV><BR></DIV><BR><FONT face=sans-serif size=2>When a SIP call starts (INVITE
/ 200 OK), asterisk seems to create a random port number for voice (rtp)
packets. Is it possible to force this port value (without using reinvite since
i am trying to use SIP against something else than sip)</FONT> <BR><BR><FONT
face=sans-serif size=2>thanks a lot in advance</FONT> <BR>
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