[asterisk-dev] Error in h323 channel

[Diego Moreno]

I done it.
http://bugs.digium.com/view.php?id=9063

Best regards.


2007/2/13, Paul Cadach <paul at odt.east.telecom.kz>:
>
>  Please, open a bug and attach call trace with next options:
> core set verbose 3
> core set debug 2
> h323 set debug
> h323 set trace 5
>
> Be sure you enabled verbose and debug options in /etc/asterisk/logger.conf
> to be able to record verbose/debug messages into log file.
>
>
> WBR,
> Paul.
>
> ----- Original Message -----
> *From:* Diego Moreno <dmoreno at dit.upm.es>
> *To:* Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> *Sent:* Monday, February 12, 2007 9:40 AM
> *Subject:* [asterisk-dev] Error in h323 channel
>
> Hi,
> I have Asterisk 1.4.0 installed in a Ubuntu Dapper. Its works as sip
> server and I am trying to get call from h323 to sip. I have a gnugk (GNU
> Gatekeeper) installed and configured. To call from h323 I use a Polycom PVX
> 8.0.0 and to receive the call I use a Ekiga Softphone registered as SIP
> user.
>
> When the call rings and I pick the softphone, the call finish with this
> error in asterisk logs:
>
> [Feb 12 17:34:16] ERROR[6311]: chan_h323.c:1848 external_rtp_create:
> Unable to find call ip$138.4.24.66:3236/13438(13438)
>
> ERROR: on_external_rtp_create failure
>
> Where could be the error?
>
> Thanks!
>
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