[Asterisk-Users] H323 Call Dropping

[Thomas Kuepper]

try this



Dial(H323/h323:${EXTEN}@gkip


Am 05.08.2004 um 14:34 schrieb 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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>
--
Thomas Küpper

01063 Telecom GmbH & Co. KG
Mottmannstr. 2
53842 Troisdorf

Telefon: 02241-9434-506
Telefax: 02241-9434-846

E-Mail: thomas.kuepper at 01063telecom.de
E-Mail: tk at teldafax.de
Homepage: http://www.01063telecom.de

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