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/[EMAIL PROTECTED]") 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. [EMAIL PROTECTED]
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
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