[Asterisk-Users] OH.323 Dialout Problem

[administrator tootai]

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