[asterisk-users] codec_g729.so not work!

Zhang Shukun bitzsk at gmail.com
Thu Aug 19 21:28:27 CDT 2010


hi, all
      i want to use g729 codec for set up a call. so i donwloaded the
so file from web site: http://asterisk.hosting.lv/#bin
and install it properly.

*CLI>
*CLI> core show translation
         Translation times between formats (in microseconds) for one
second of data
          Source Format (Rows) Destination Format (Columns)

           g723   gsm  ulaw  alaw g726aal2 adpcm  slin lpc10  g729
speex  ilbc  g726  g722 siren7 siren14 slin16
     g723     -     -     -     -        -     -     -     -     -
-     -     -     -      -       -      -
      gsm     -     -     2     2     2000     2     1  3001  3000
-     -  2001  1001      -       -   2001
     ulaw     -  3000     -     1     2000     2     1  3001  3000
-     -  2001  1001      -       -   2001
     alaw     -  3000     1     -     2000     2     1  3001  3000
-     -  2001  1001      -       -   2001
 g726aal2     -  3999  1001  1001        -  1001  1000  4000  3999
-     -  3000  2000      -       -   3000
    adpcm     -  3999  1001  1001     2999     -  1000  4000  3999
-     -  3000  2000      -       -   3000
     slin     -  2999     1     1     1999     1     -  3000  2999
-     -  2000  1000      -       -   2000
    lpc10     -  4998  2000  2000     3998  2000  1999     -  4998
-     -  3999  2999      -       -   3999
     g729     -  3999  1001  1001     2999  1001  1000  4000     -
-     -  3000  2000      -       -   3000
    speex     -     -     -     -        -     -     -     -     -
-     -     -     -      -       -      -
     ilbc     -     -     -     -        -     -     -     -     -
-     -     -     -      -       -      -
     g726     -  3998  1000  1000     2998  1000   999  3999  3998
-     -     -  1999      -       -   2999
     g722     -  3998  1000  1000     2998  1000   999  3999  3998
-     -  2999     -      -       -   1000
   siren7     -     -     -     -        -     -     -     -     -
-     -     -     -      -       -      -
  siren14     -     -     -     -        -     -     -     -     -
-     -     -     -      -       -      -
   slin16     -  4998  2000  2000     3998  2000  1999  4999  4998
-     -  3999  1000      -       -      -

my sip.conf add two account:

[123]
type=friend
username=123
host=dynamic
context=95040
dtmfmode=rfc2833
disallow=all
allow=g729
insecure=port,invite
canreinvite=no
nat=yes

[321]
type=friend
username=321
host=dynamic
context=95040
dtmfmode=rfc2833
disallow=all
allow=g729
insecure=port,invite
canreinvite=no

my extension is :

exten => 321,1,Dial(SIP/321)


when i want to set up a call (123 dial 321). but failed. it says:

 == Using SIP RTP CoS mark 5
[Aug 20 18:23:21] NOTICE[14543]: chan_sip.c:8454 process_sdp: No
compatible codecs, not accepting this offer!

Could you tell me what 's wrong?


-- 
Thanks & Regards
Sucan



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