[Asterisk-Users] H323 Call Dropping

[Asterisk .]

Hello All,

I am trying to setup a SIP to H323 system using SER, Asterisk And GnuGK. Following is the
configuration:

CISCO ATA (NAT) -> SER -> ASTERISK -> GNUGK

My Cisco ATA is registered with SER and When I dial a number, SER forwards the call to Asterisk,
and Asterisk forwards the call to the GateKeper. This is ok, call reaches the gatekeeper, however
the gatekeeper drops the call immediately after receiving it. Can anyone tell me what is the
reason for this? Is it a codec issue or anything i have misconfigured? I would sincerly appreciate
any help or guidence on this. I am using Nufone Network's chan_h323 driver.  

This is from the Asterisk console:

-- Executing Dial("SIP/XXX.XX.XXX.XXX-080f5e78", "h323/h323:14083339452 at XX.XX.XXX.XX") in new
stack
 -- Called h323:14083339452 at XX.XX.XXX.XX
 == No one is available to answer at this time
 -- Executing Hangup("SIP/XXX.XX.XXX.XXX-080f5e78", "") in new stack
== Spawn extension (default, 14083339452, 2) exited non-zero on 'SIP/XXX.XX.XXX.XXX-080f5e78'


This is the gatekeeper log:
ACF|XXX.XX.XXX.XXX:1723|3950_endp|5285|14083339452:dialedDigits|995041321:dialedDigits|false;
DCF|XXX.XX.XXX.XXX|3950_endp|5285|normalDrop;

Registration details on gatekeeper for Asterisk:
?
AllRegistrations
RCF|XXX.XX.XXX.XXX:1723|root:h323_ID|gateway|3950_endp

This is from h323.conf:

[general]
port = 1723
disallow=all
allow=g723.1
allow=ulaw
allow=alaw
allow=gsm

This is from sip.conf:

[general]
context=default
port=5070
disallow=all
allow=g723.1
allow=ulaw
allow=alaw
allow=ilibc
allow=gsm

Extensions.conf has these entries in the default context:
exten => _.,1,Dial(h323/h323:${EXTEN}@gatekeeper_ip)
exten => _.,2,Hangup

*CLI> show version
Asterisk 1.0-RC1 built by root at localhost on a i686 running Linux

TIA...

/G


		
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