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

harrygaillac-sip at yahoo.fr harrygaillac-sip at yahoo.fr
Fri Sep 8 02:00:56 MST 2006


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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