[asterisk-users] One Way Audio Problem
GNUbie
gnubie at gmail.com
Wed Oct 15 20:18:46 CDT 2008
Hello,
On Tue, Oct 14, 2008 at 12:07 AM, Jeff LaCoursiere <jeff at jeff.net> wrote:
>
> A packet trace will probably show exactly what is happening. Try:
>
> tcpdump -nlXs 8192 -i eth0 port 5060
>
> You should be able to see the SIP information going back and forth and
> will probably show you that your NAT rules are applying when they
> shouldn't. I agree with first turning off your firewall and testing...
> but if that actually solves the problem you need to know why. This should
> tell why.
Why eth0 when in fact it is not being used AFAIK? My Asterisk box is
connected to the LAN via its eth1 interface and the SIP phone is
calling from the LAN to the analog telephone via FXO/POTS. Again,
below is the call scenario diagram:
[SNOM] ==LAN==> eth1 [ASTERISK] fxo ==POTS==> [ANALOG_TELEPHONE]
eth0
||
INTERNET
Please advice. Thank you in advance.
Regards,
GNUbie
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