asterisk users - Feb 2007 - Asterisk & CME integration using h323

Hi all, I'm trying to trunk my asterisk with a cisco cme 3.3 using h323.

Cisco conf:

dial-peer voice 8 voip
  destination-pattern 2...
  session target ipv4:<asterisk ip>
  codec g711alaw
  no vad

h323.conf

[general]
port = 1720
bindaddr = 0.0.0.0
;tos=lowdelay
;
disallow=all
allow=alaw
allow=ulaw
allow=gsm
context=from-internal

extension.conf

[from-internal]

exten => _1XXX,1,Dial(SIP/${EXTEN}@<cme ip>)
exten => 2000,1,Dial(SIP/2000)

I'm able from Asterisk to call ip phone connected to cme but from cme to 
asterisk the phones ring but go in hangup immediatly.

My debug:

---
localhosAnswering call ip$192.168.99.2:53716/21
localhos-- Transmitting RFC2833 on payload 101
localhos-- Received Facility message...
localhos-- Received Facility message...
localhos-- Inbound RFC2833 on payload 101
localhos-- Received RELEASE COMPLETE message...
localhos-- ClearCall: Request to clear call with token 
ip$192.168.99.2:53716/21, cause 22
localhos-- Sending RELEASE COMPLETE
localhost*CLI>  channelsOpen = 1
                 channelsOpen = 0
localhos-- ClearCall: Request to clear call with token 
ip$192.168.99.2:53716/21, cause 7
Scheduling destruction of call 
'7d299d880a84eea37f6da0c10b26b2b2@192.168.99.254' in 32000 ms
set_destination: Parsing <sip:2000@192.168.99.122:5060> for address/port 
to send to
set_destination: set destination to 192.168.99.122, port 5060
Reliably Transmitting (no NAT) to 192.168.99.122:5060:
BYE sip:2000@192.168.99.122:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.99.254:5060;branch=z9hG4bK1a8c7be3;rport
From: "1003" <sip:1003@192.168.99.254>;tag=as769a2c55
To: <sip:2000@192.168.99.122:5060>;tag=1473512925
Contact: <sip:1003@192.168.99.254>
Call-ID: 7d299d880a84eea37f6da0c10b26b2b2@192.168.99.254
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
localhosExternalRTPChannel Destroyed
localhosExternalRTPChannel Destroyed
  -- Call with Enrico [192.168.99.2] completed (22)
Feb 14 11:36:41 NOTICE[31439]: chan_h323.c:1479 cleanup_connection: 
Avoiding H.323 destory deadlock on ip$192.168.99.2:53716/21
localhost*CLI>
<-- SIP read from 192.168.99.122:5060:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.99.254:5060;branch=z9hG4bK1a8c7be3;rport
From: "1003" <sip:1003@192.168.99.254>;tag=as769a2c55
To: <sip:2000@192.168.99.122:5060>;tag=1473512925
Contact: <sip:2000@192.168.99.122:5060>
Call-ID: 7d299d880a84eea37f6da0c10b26b2b2@192.168.99.254
CSeq: 103 BYE
Server: X-Lite release 1105d
Content-Length: 0


--- (9 headers 0 lines) ---
Destroying call '7d299d880a84eea37f6da0c10b26b2b2@192.168.99.254'
Feb 14 11:36:41 NOTICE[31439]: chan_h323.c:1479 cleanup_connection: 
Avoiding H.323 destory deadlock on ip$192.168.99.2:53716/21
Feb 14 11:36:41 NOTICE[31439]: chan_h323.c:1479 cleanup_connection: 
Avoiding H.323 destory deadlock on ip$192.168.99.2:53716/21
Feb 14 11:36:41 NOTICE[31439]: chan_h323.c:1479 cleanup_connection: 
Avoiding H.323 destory deadlock on ip$192.168.99.2:53716/21
localhos== H.323 Connection deleted.


I don't understand why the call goes down only from cisco to 
asterisk.... any ideas?


Thanks Enrico
-- 
Pasqualotto 'Pasqu' Enrico
enrico AT pasqualotto DOT org
web: http://www.pasqualotto.org
skype: epasqualotto