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

[Sebastian]

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
>