[Asterisk-Users] H.323 - SIP gateway

[Olaf Menzel]

I am trying to configure * to route calls from SIP extension to an 
externeal H.323 gatekeeper and vice versa.
The route from * to the gatekeeper is a simple ENUM call and work fine:

[outbound][outbound]
exten => _3XXX,1,Dial,H323/${EXTEN}@10.3.1.100

One Snom100 phone is defined in sip.conf:

[snom]
type=friend
host=dynamic
dtmfmode=rfc2833
mailbox=1000
context=local
callerid="Operator Office" <1000>

in extensions.conf it is defined as extension 1000:

[local]
include => voicemail
include => outbound
include => inbound

; SIP Phone Operator Office
exten => 1000,1,Dial,SIP/snom|30|tr
exten => 1000,2,Voicemail,u1000
exten => 1000,102,Voicemail,b1000


When I  call a H.323 telephone  behind the gatekeeper this  telephone 
shows a callerid "root" as name and the Asterisk IP address without the 
original extension 1000. I can define an alias in h323.conf but every 
call to the gatekeeper will have this callerid:

 [olaf-snom]
 type=h323
 e164=1000
 context=local

But I want to transmit  the original callerid as defined in sip.conf via 
the H.323 gatekeeper to a H.322 phone. How to manage this ??

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I have defined a inbound gatekeeper in h323.conf:

[Corponet]
type=user
host=10.3.1.100
context=inbound
incominglimit=4

What else has to be in the extensions.conf if a H.323 phone want to call 
me at: 1000 at asteriskIPAddress ??

regards

Olaf