On Wed, Feb 1, 2012 at 1:16 PM, Ahmed Munir <ahmedmunir...@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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