First at all, thanks guys for the support!!<br><br>I've been doing what people told me. To asure that I have DirectX on SJPhone (audio setting option enable DirectX 8.1) and I<br>can't run fxotune because I don't use this cards (sorry if I'm wrong). I'm just trying to probe my * box with the
<a href="http://voip-info.org">voip-info.org</a><br>tutorials. <br><br>I turn down the mic gain and the pc's has the enough power. I make the echo test on the 2 clients (dialing 500 on *) and it sounds<br>great (fluid voice).
<br><br>This is my [general]sip.conf format: <br><br>I omitted other parts which were on comments because are examples from the web site<br><br><br>[general]<br>context=default<br>;allowguest=no <br>
<br>;realm=mydomain.tld <br>bindport=5060<br>bindaddr=<a href="http://0.0.0.0">0.0.0.0</a><br>srvlookup=yes <br>;domain=mydomain.tld<br>;************** Cambio de lineas<br>disallow=all<br>;allow=g729<br>allow=gsm<br>allow=ulaw
<br>jitterbuffer=yes<br>maxjitterbuffer=800<br>;allow=ilbc<br>;musicclass=default<br>;language=en<br>;relaxdtmf=yes<br><br>rtptimeout=60<br> <br>;rtpholdtimeout=300<br><br>;trustrpid = no<br>;sendrpid = yes<br>
;progressinband=never<br> <br>;useragent=Asterisk PBX<br>;promiscredir = no <br> <br>;usereqphone = no <br><br>;*********** Cambio de lineas DTMFMODE estaba en comentarios ********************
<br><br>dtmfmode = rfc2833 <br>;compactheaders = yes <br>;sipdebug = yes <br> <br>;subscribecontext = default <br> <br>
<br>;notifyringing = yes <br><br>;******************** Usuario 1 ************************<br>[usuario1]<br>type=friend<br> host=dynamic<br> dtmfmode=rfc2833<br> username=usuario1
<br> secret=usuario1<br><br>;******************** Usuario 2 ************************<br>[usuario2]<br>type=friend<br> host=dynamic<br> dtmfmode=rfc2833<br> username=usuario2<br> secret=usuario2<br><br><br>And again thanks for the help!
<br><br><br>Carlos Bernat<br><br>