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

[Sebastian]

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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