[Asterisk-Users] OH.323 Dialout Problem

[Brian Wilkins]

Hmm, well my gatekeeper only supports G723 and according to the Asterisk Wiki:
http://www.voip-info.org/wiki-Asterisk+G.723+pass-thru

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 écrit :
> >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/##########@xxx.xxx.xxx.xx: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.

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