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.Hi all, I am setting up a a proof on concept where a SIP phone sits on the internet and connects to a * behing a NAT. Right now the SIP phone connects to the * box just fine, I can dial and I see the commands being executed on the * box, but I don't have any audio on the SIP phone. Any idas/pointers?
-- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 |
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