Brian Wilkins a �crit :
Hmm, well my gatekeeper only supports G723 and according to the Asterisk Wiki:
http://www.voip-info.org/wiki-Asterisk+G.723+pass-thru
Nope. Only codec I know to work with my GnuGK EP (GnomeMeeting) are
g711ulaw&alaw and gsm Dont know with oh323, but in nufone h323 you can
see activated codecs with h.323 show codecs
G723 is supported in pass-thru mode. I am placing a SIP to H323 call, so if I
understand it right, it should work since I am working in pass-thru mode.
On Friday 13 August 2004 08:10 pm, administrator tootai wrote:
Brian Wilkins a �crit :
Hi,
I am using the Grandstream HandyTone 486 as a SIP Adapter with a
regular phone. Asterisk configuration is listed below. When I attempt to
place a H.323 call, I receive the following errors:
- Executing Dial("SIP/2000-3029",
"OH323/[EMAIL PROTECTED]:1720|20") in new stack
Aug 13 09:13:03 WARNING[20497]: channel.c:1806 ast_request: No translator
path exists for channel type OH323 (native 1) to 4
Aug 13 09:13:03 NOTICE[20497]: app_dial.c:705 dial_exec: Unable to create
channel of type 'OH323'
== Everyone is busy at this time
-- Executing Congestion("SIP/2000-3029", "") in new stack
== Spawn extension (default, ##########, 2) exited non-zero on
'SIP/2000-3029'
The Grandstream HandyTone is registered as SIP extension 2000. The
Grandstream HandyTone is configured to use the codec G723 6.3 with 32
frames.
Codec issue. Asterisk doesn't support g723. Try g711 instead.
--
Daniel
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