[Asterisk-Users] H323 Call Dropping

[Sebastian Nocetti]

Dial(h323/h323:${EXTEN}@gatekeeper_ip)

I think problem is in this line...

Dial(h323/${EXTEN}@gatekeeper_ip) 

That's not the correct way?

-----Mensaje original-----
De: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] En nombre de Asterisk .
Enviado el: Jueves, 05 de Agosto de 2004 07:07 a.m.
Para: asterisk-users at lists.digium.com
Asunto: Re: [Asterisk-Users] H323 Call Dropping

Can anyone please help?

--- "Asterisk ." <asterisk_in at yahoo.com> wrote:

> 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|995041
> ACF|321: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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