[Asterisk-Users] OH.323 Dialout Problem

[administrator tootai]

Brian Wilkins a écrit :

>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 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.
>>    
>>
>
>  
>
-- 
Daniel