[Asterisk-Users] [Serusers] No translator path exists for channel type H323 :-| :-| :-|

[Carlos Maynard]

Hello there,

I was playing with my Asterisk lastnite and was able to terminate calls 
from the console, or from a SIP Phone (ATA 186) to my H323 carriers (2 
of them). After a few seconds connected the call would get disconnected 
and no audio was ever heard back and forth.

ANYWAYS, the thing is i wake up today to try and continue with my 
tests... and it turns out that it simply doesn't work now. I don't have 
my ATA186 handy now (I'm at my office), but i'm trying to call from the 
console and i'm getting the following error:

rrcs-67-79-16-43*CLI> dial 8888
    -- Executing Dial("OSS/dsp", "H323/18329287763 at 61.101.71.143") in 
new stack
Apr  7 09:56:33 WARNING[393238]: channel.c:1838 ast_request: No 
translator path exists for channel type H323 (native 256) to 64
Apr  7 09:56:33 NOTICE[393238]: app_dial.c:714 dial_exec: Unable to 
create channel of type 'H323'
  == Everyone is busy/congested at this time
Apr  7 09:56:43 WARNING[393238]: pbx.c:1924 ast_pbx_run: Timeout, but no 
rule 't' in context 'local'

I've been looking at the mailing list but can't find posts with the same 
error.

Any ideas/suggestions will be greatly appreciated.

Thanks in advance,

Carlos Maynard jr.
carlos at hottelephone.com


P.S.: i'm running Asterisk 1.0-RC2... here is my h323.conf:

;/***************************************************
;h323.conf
;/***************************************************
[general]
port = 1720
bindaddr = 0.0.0.0

disallow=all
allow=g729
gatekeeper = 61.19.16.45
alias = carlasterisk
AllowGKRouted = yes

[asterisk]
type=h323
prefix=111
context=local