[asterisk-dev] Error in h323 channel

[Paul Cadach]

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 
  To: Asterisk Developers Mailing List 
  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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