[asterisk-users] one way audio for xlite clients behind NAT

[Thomas Johnson]

the client that is behind nat is
[tomfmason]
type=friend
secret=secret
callerid="Thomas Johnson" <XXXX>
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=gsm
allow=ulaw
allow=alaw
qualify=yes
context=sip

do I have to enable nat on all of them?
On Thu, Sep 16, 2010 at 1:36 PM, Sebastian <shop at open-t.co.uk> wrote:

>
>
> On 09/16/2010 06:58 PM, Thomas Johnson wrote:
> > I am having a one way audio issue with xlite clients behind NAT. They
> > can connect to the server and make calls but no audio is heard on the
> > other end.
> >
> > my sip conf
> >
> > [general]
> > context=default
> > bindport=5060
> > bindaddr=0.0.0.0
> > srvlookup=yes
> > canreinvite=no
> >
> > [tomfmason]
> > type=friend
> > secret=secret
> > callerid="Thomas Johnson"  <XXXX>
> > host=dynamic
> > nat=yes
> > canreinvite=no
> > disallow=all
> > allow=gsm
> > allow=ulaw
> > allow=alaw
> > qualify=yes
> > context=sip
> >
> > [1001];Work
> > type=peer
> > dtmfmode=rfc2833
> > context=sip
> > insecure=very
> > host=sip.domain.com  <http://sip.domain.com>
> > nat=no
> >
> > [1000];IPKall
> > type=peer
> > dtmfmode=rfc2833
> > context=sip
> > insecure=very
> > host=voiper.ipkall.com  <http://voiper.ipkall.com>
> > nat=no
>
> You seem to be using nat=no
>
> shouldn't that be nat=yes?
>
> >
> >
> >
> > I pasted the log here ->  http://pastie.org/1163238
> >
> >
> > I have tried connecting both of the clients to another sip service(
> sip2sip.info  <http://sip2sip.info>) and did not have the same problems.
> >
> >
> > Any suggestions would be great.
> >
> > Thanks,
> >
> > Tom
> >
>
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