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On Mon, 2005-11-14 at 11:57 -0500, Andre Courchesne - Consultant wrote:
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<FONT COLOR="#000000">Hi all,</FONT>
<FONT COLOR="#000000"> I am setting up a a proof on concept where a SIP phone sits on the </FONT>
<FONT COLOR="#000000">internet and connects to a * behing a NAT.</FONT>
<FONT COLOR="#000000"> Right now the SIP phone connects to the * box just fine, I can dial </FONT>
<FONT COLOR="#000000">and I see the commands being executed on the * box, but I don't have any </FONT>
<FONT COLOR="#000000">audio on the SIP phone. Any idas/pointers?</FONT>
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That usually means that you have not forwarded the RTP ports. By default Asterisk uses ports 10000 - 20000 for RTP on UDP so you need to tell your NAT device to forward those ports to your * box.<BR>
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--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001
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