On Nov 14, 2005, at 11:57 AM, Andre Courchesne - Consultant wrote:
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?
I would recommend that you do a little research on google, voip-
info.org, and the list archives.
To connect to an Asterisk box that sits behind NAT, you need to
forward ports 5060 and 10000-20000 too the asterisk box, and you need
to configure the externip, localnet, and nat variables in sip.conf.
audio problems are almost always due to the RTP stream (ports
10000-20000) not being forwarded properly, either due to the port
forwarding setup or the sip.conf settings.
Tom
----------------------------------------------------------
Tom Rymes
Cascade Link Systems
www.cascadelinksystems.com
(603) 375-1414
Technology solutions for small and medium sized businesses.
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