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
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 voip-info.org
tutorials.
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
great (fluid voice).
This is my [general]sip.conf format:
I omitted other parts which were on comments because are examples from the web site
[general]
context=default
;allowguest=no
;realm=mydomain.tld
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
;domain=mydomain.tld
;************** Cambio de lineas
disallow=all
;allow=g729
allow=gsm
allow=ulaw
jitterbuffer=yes
maxjitterbuffer=800
;allow=ilbc
;musicclass=default
;language=en
;relaxdtmf=yes
rtptimeout=60
;rtpholdtimeout=300
;trustrpid = no
;sendrpid = yes
;progressinband=never
;useragent=Asterisk PBX
;promiscredir = no
;usereqphone = no
;*********** Cambio de lineas DTMFMODE estaba en comentarios ********************
dtmfmode = rfc2833
;compactheaders = yes
;sipdebug = yes
;subscribecontext = default
;notifyringing = yes
;******************** Usuario 1 ************************
[usuario1]
type=friend
host=dynamic
dtmfmode=rfc2833
username=usuario1
secret=usuario1
;******************** Usuario 2 ************************
[usuario2]
type=friend
host=dynamic
dtmfmode=rfc2833
username=usuario2
secret=usuario2
And again thanks for the help!
Carlos Bernat
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