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

[Thomas Johnson]

I have tried doing that with just ulaw and alaw, respectively, and nothing
changed

Also, if I disable the firewall in my router I lose incoming audio and
outgoing audio works.



On Thu, Sep 16, 2010 at 2:50 PM, Sebastian <shop at open-t.co.uk> wrote:

>
>
> On 09/16/2010 07:59 PM, Thomas Johnson wrote:
> > 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?
>
> I don't think so. It's just that you didn't specify which client is which.
>
> My next guess is that it is a codec problem. I've had similar problems -
> and upon checking Asterisk logs - I discovered that the client and
> Asterisk weren't agreeing correctly on codecs. Can you double-check your
> X-lite configuration - and maybe try to ulaw or alaw as the only codec
> at both ends?
>
> Sebastian
>
> > On Thu, Sep 16, 2010 at 1:36 PM, Sebastian <shop at open-t.co.uk
> > <mailto: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> <
> http://sip.domain.com>
> >      > nat=no
> >      >
> >      > [1000];IPKall
> >      > type=peer
> >      > dtmfmode=rfc2833
> >      > context=sip
> >      > insecure=very
> >      > host=voiper.ipkall.com <http://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> <http://sip2sip.info>)
> >     and did not have the same problems.
> >      >
> >      >
> >      > Any suggestions would be great.
> >      >
> >      > Thanks,
> >      >
> >      > Tom
> >      >
> >
> >     --
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