[asterisk-users] codecs translation in Asterisk SVN-trunk-r41990 [SOLVED]

harrygaillac-sip at yahoo.fr harrygaillac-sip at yahoo.fr
Fri Sep 8 02:19:32 MST 2006


--- harrygaillac-sip at yahoo.fr a écrit :

> Hello,
> 
> I recorded some files (gsm format) but i can not
> hear these files without g729 
>
--------------------------------------------------------
>     -- Executing [84 at sip:1]
> Answer("SIP/86-08218198",
> "") in new stack
>     -- Executing [84 at sip:2] Dial("SIP/86-08218198",
> "Sip/84|30|tTj") in new stack
>     -- Called 84
>     -- Got SIP response 486 "Busy Here" back from
> 80.119.15.85
>     -- SIP/84-0821ddd0 is busy
>   == Everyone is busy/congested at this time
> (1:1/0/0)
>     -- Executing [84 at sip:3]
> VoiceMail("SIP/86-08218198", "u84") in new stack
>     -- Playing 'vm-theperson' (language 'fr')
> [2006-09-08 10:38:43] WARNING[2382]: channel.c:2609
> set_format: Unable to find a codec translation path
> from g729 to gsm
> [2006-09-08 10:38:43] WARNING[2382]: file.c:805
> ast_streamfile: Unable to open digits/8 (format
> 0x100
> (g729)): No such file or directory
>   == Spawn extension (sip, 84, 3) exited non-zero on
> 'SIP/86-08218198'
> serveur1*CLI> show v
> version    video      voicemail
> serveur1*CLI> show version
> Asterisk SVN-trunk-r41990 built by root @
> serveur1.home.net on a i686 running Linux on
> 2006-09-04 17:07:12 UTC
> serveur1*CLI>
> 
>
--------------------------------------------------------
> 
> 
> 
> 
> 	
> 
> 	
> 		
>
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___________________________________________________________________________ 
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