Hi,
I’m newbie in asterisk and voip, but I have a half success. I have a an asterisk on a debian lenny to which a connect with a softwarephone.
this asterisk connect to an avaya pbx with a h323 trunk. If there is incoming call I can answer with softwarephone. My question is how can I route all outgoing call to the trunk to avaya? If somebody can send me a sample extension config I will be very happy 
My configs:
h323.conf
[general]
port = x
bindaddr = x.x.x.x
progress_setup = 8
progress_alert = 8
faststart=yes
h245tunneling=yes
gatekeeper = DISABLE
;We need to conserve the main parameters to allow the h323 to call to the SIP phone
disallow=all
allow=alaw
dtmfmode=inband
context=internal ; name of your context
[Avaya]
type=friend
context=internal
host=x.x.x.x; IP Address of your CLAN
port=x; port used to connect on CLAN it could be some others port regarding your configuration in signalling grou$
disallow=all
allow=alaw
canreinvite=no
dtmfmode=inband
sip.conf
[general]
context=internal
bindaddr=x.x.x.x; IP Address of your Asterisk
srvlookup=yes
videosupport=no ; if you want activate video support
canreinvite=no
[59000]
type=friend
username=59000 ; for xlite
secret=password ;your password
host=dynamic
dtmfmode=inband
disallow=all
allow=ulaw ; alaw
; allow=h263 ; to use a video codec if needed
callerid=“xxxxx” <59000>
nat=yes
context=internal
extensions.conf
[general]
static=yes
autofallthrough=yes
[internal]
;My extension 59xxx
exten => 59000,1,Dial(SIP/59000)
exten => 59000,2,HangUp()
[outgoing]
this is the question!!
Thanks,
cyd