[asterisk-users] Getting one way audio even NAT is configured

[Warren Selby]

On Wed, Feb 1, 2012 at 1:16 PM, Ahmed Munir <ahmedmunir007 at gmail.com> wrote:

> Hi all,
>
> I'm getting one way audio when calling over the SIP trunk i.e. end device
> B (remote end of SIP trunk) can hear device A (softphone registered with
> Asterisk) but device A can't hear device B. Even though I configured same
> NAT configurations on other servers and they are working good. The NAT
> configuration is listed below;
>
> localnet=130.0.0.0/130.0.0.0
> externhost=12.131.12.13
> externrefresh=10
> fromdomain=test.localhost.com
> nat=yes
> qualify=yes
> canreinvite=no
>
>
> NAT on device end i.e. my softphone (extension) has already set to yes
> with canreinvite=no  but still unable to resolve this issue. SIP traces are
> listed below;
>
>
<snip>


>
> The Asterisk version I'm using is 1.8.5. Please assist me at earliest.
>

Which device (A or B) is behind NAT with regards to your asterisk server?
Is that the actual localnet= statement you're using, because to my
understanding that is not the proper format to use (should be
localnet=x.x.x.x/y.y.y.y where x.x.x.x is your actual local network, and
y.y.y.y is your subnet for your local network).

-- 
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com <http://www.selbytech.com>
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