<div>I have tried doing that with just ulaw and alaw, respectively, and nothing changed</div><div><br></div>Also, if I disable the firewall in my router I lose incoming audio and outgoing audio works.<div><br></div><div><br>
<br><div class="gmail_quote">On Thu, Sep 16, 2010 at 2:50 PM, Sebastian <span dir="ltr"><<a href="mailto:shop@open-t.co.uk">shop@open-t.co.uk</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div class="im"><br>
<br>
On 09/16/2010 07:59 PM, Thomas Johnson wrote:<br>
> the client that is behind nat is<br>
> [tomfmason]<br>
> type=friend<br>
> secret=secret<br>
> callerid="Thomas Johnson" <XXXX><br>
> host=dynamic<br>
> nat=yes<br>
> canreinvite=no<br>
> disallow=all<br>
> allow=gsm<br>
> allow=ulaw<br>
> allow=alaw<br>
> qualify=yes<br>
> context=sip<br>
><br>
> do I have to enable nat on all of them?<br>
<br>
</div>I don't think so. It's just that you didn't specify which client is which.<br>
<br>
My next guess is that it is a codec problem. I've had similar problems -<br>
and upon checking Asterisk logs - I discovered that the client and<br>
Asterisk weren't agreeing correctly on codecs. Can you double-check your<br>
X-lite configuration - and maybe try to ulaw or alaw as the only codec<br>
at both ends?<br>
<br>
Sebastian<br>
<div class="im"><br>
> On Thu, Sep 16, 2010 at 1:36 PM, Sebastian <<a href="mailto:shop@open-t.co.uk">shop@open-t.co.uk</a><br>
</div><div><div></div><div class="h5">> <mailto:<a href="mailto:shop@open-t.co.uk">shop@open-t.co.uk</a>>> wrote:<br>
><br>
><br>
><br>
> On 09/16/2010 06:58 PM, Thomas Johnson wrote:<br>
> > I am having a one way audio issue with xlite clients behind NAT. They<br>
> > can connect to the server and make calls but no audio is heard on the<br>
> > other end.<br>
> ><br>
> > my sip conf<br>
> ><br>
> > [general]<br>
> > context=default<br>
> > bindport=5060<br>
> > bindaddr=0.0.0.0<br>
> > srvlookup=yes<br>
> > canreinvite=no<br>
> ><br>
> > [tomfmason]<br>
> > type=friend<br>
> > secret=secret<br>
> > callerid="Thomas Johnson" <XXXX><br>
> > host=dynamic<br>
> > nat=yes<br>
> > canreinvite=no<br>
> > disallow=all<br>
> > allow=gsm<br>
> > allow=ulaw<br>
> > allow=alaw<br>
> > qualify=yes<br>
> > context=sip<br>
> ><br>
> > [1001];Work<br>
> > type=peer<br>
> > dtmfmode=rfc2833<br>
> > context=sip<br>
> > insecure=very<br>
</div></div>> > host=<a href="http://sip.domain.com" target="_blank">sip.domain.com</a> <<a href="http://sip.domain.com" target="_blank">http://sip.domain.com</a>> <<a href="http://sip.domain.com" target="_blank">http://sip.domain.com</a>><br>
<div class="im">> > nat=no<br>
> ><br>
> > [1000];IPKall<br>
> > type=peer<br>
> > dtmfmode=rfc2833<br>
> > context=sip<br>
> > insecure=very<br>
> > host=<a href="http://voiper.ipkall.com" target="_blank">voiper.ipkall.com</a> <<a href="http://voiper.ipkall.com" target="_blank">http://voiper.ipkall.com</a>><br>
> <<a href="http://voiper.ipkall.com" target="_blank">http://voiper.ipkall.com</a>><br>
> > nat=no<br>
><br>
> You seem to be using nat=no<br>
><br>
> shouldn't that be nat=yes?<br>
><br>
> ><br>
> ><br>
> ><br>
> > I pasted the log here -> <a href="http://pastie.org/1163238" target="_blank">http://pastie.org/1163238</a><br>
> ><br>
> ><br>
> > I have tried connecting both of the clients to another sip<br>
</div>> service(<a href="http://sip2sip.info" target="_blank">sip2sip.info</a> <<a href="http://sip2sip.info" target="_blank">http://sip2sip.info</a>> <<a href="http://sip2sip.info" target="_blank">http://sip2sip.info</a>>)<br>
<div class="im">> and did not have the same problems.<br>
> ><br>
> ><br>
> > Any suggestions would be great.<br>
> ><br>
> > Thanks,<br>
> ><br>
> > Tom<br>
> ><br>
><br>
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