[Asterisk-Users] Connection of Asterisk to Cisco Callmanager via H.323

[Paul Davidson]

Greetings-

I've managed to successfully integrate our Cisco Callmanager (v3.3.3)
to Asterisk for the purposes of replacing a broken and expensive
meetme system with one that works.  Under CCM v3.3.3, there is no
support for SIP trunks- our 4.0 migration is still some time off.  I'm
using the GnuGK Open Gatekeeper to handle the signalling, and I've got
an H.323 trunk set up.  Calls from the Cisco side into Asterisk work
swimmingly- very smooth. (kudos to Mark).  I'm using the NuFone H.323
channel driver, and OH323 1.12.2.  I'm running Asterisk 1.0 RC2.

However, here's the rub.  For some reason, I cannot get calls back
from Asterisk to the Callmanager.  If I use OhPhone pointed directly
at the gatekeeper, I can get it to work- so I've ruled out Callmanager
or Gatekeeper setups- but calls from Asterisk to Callmanager (using
DIAX or other softphone authenticated to Asterisk) fail- the softphone
keeps ringing, but the phone doesn't.  DIAX will make successful calls
into Asterisk for other numbers, so I've got the IAX2 authentication
apparently working- but there's a problem somewhere (I believe) in the
signalling from Asterisk to Callmanager.  Note that I've got the
prefix 781 set up to route to the Callmanager- feed 781XXXX to the
gatekeeper via OhPhone, and extension XXXX rings on my desk.  From
tcpdump and ethereal, I'm seeing a Q.931 SETUP message going back to
the Callmanager from Asterisk when I attempt the call- it seems to
have the correct information, but differs in some key ways from the
working OhPhone call- some IE's are missing (calling party, for
instance- though called party is there and correct), and the Bearer
Capability is set to 'Speech' out of Asterisk, and 'Unrestricted
digital information' out of OhPhone.

I've included relevant snippets of my h323.conf and extensions.conf
below- has anyone sucessfully done this?  Once I get it 100% working,
I'll fill in the details back to the Wiki- there appears to be a
working example based on SIP there already, with the ominous note
'Upgrade to 1.0 and it will work under H.323 too!'.  Not very helpful,
I'm afraid.

Thanks in advance- snippets follow.
-Paul Davidson

------ snip h323.conf sanitized for your protection - IP address
replaced with XX------
[general]
;
; Port to listen to
port=1720
bindaddr = XX.XX.XX.105
gatekeeper=XX.XX.XX.112
disallow=all
allow=ulaw
context=voip-h323
;
; Specify alias(es) of this host.
; It may be used multiple times.
;
alias=AST
;alias=4489
;
; Set the context of H323 calls
;
[conference]
type=h323
e164=4488
context=voip-h323

------ snip extensions.conf ----------
[default]
;
include => voip-h323
; Dial pattern to get back to Cisco Callmanager
exten => _781XXXX,1,Dial(H323/${EXTEN}@192.168.222.1)
exten => _781XXXX,2,Congestion