[asterisk-users] Still voice with echo

[Carlos Alberto Bernat Orozco]

First at all, thanks guys for the support!!

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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060725/c49e6f0d/attachment.htm