[Asterisk-Users] Asterisk+GNOMEMeeting=No Sound.

[Andres Gómez García]

Hi all!

I'm newie to asterisk and I've been trying to make it work in order to
use it with Linux softphones (H.323, SIP or IAX, I don't mind) and none
hardware phone.

I'm using asterisk packages from Debian SID (my distribution), asterisk,
asterisk-config, asterisk-sounds, asterisk-h323. I've still not tried
with any IAX softphone (gnophone?) but with linphone (SIP) I've not luck
(oRTP errors in console) even to p2p connection between 2 linphone
client computers or sipomatic.

I've tried GNOMEMeeting also. It works fine with a P2P client
connections (ALSA works fine) but, even when I success connecting to an
asterisk server, I haven't hear anything. I mean, I don't hear the demo
successfull messages. I've looking the GNOMEMeeting logs and it says
that it closes the sound channel as soon as it connects to the asterisk
server. This is my h323.conf file:

[general]
port = 1720
bindaddr = 0.0.0.0

allow=all               ; turns on all installed codecs

context=default

and my extensions.conf file:

[general]
static=yes
writeprotect=no

[globals]
CONSOLE=Console/dsp
IAXINFO=guest
TRUNK=Zap/g2
TRUNKMSD=1

.
.
.

[demo]
exten => s,1,Wait,1
exten => s,2,Answer
exten => s,3,DigitTimeout,5
exten => s,4,ResponseTimeout,10
exten => s,5,BackGround(demo-congrats)
exten => s,6,BackGround(demo-instruct)

.
.
.

[default]
include => demo

.
.
.

I've also can see how asterisk says it actually plays these sound files
in the CLI.

Any idea?

Thanks in advance.
-- 
Andrés Gómez García
Ingeniero en Informática
Telf:  +34 981 91 39 91
Fax:   +34 981 91 39 49
mailto:agomez at igalia.com
http://personales.igalia.com/agomez
IGALIA, S.L. http://www.igalia.com