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
----- "Zhang Shukun" <bitzsk at gmail.com> wrote:> > == 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?<sarcasm>Besides the use of an illegal G.729 implementation? </sarcasm> Are both endpoints G.729 compatible? Most hard phones are G.729 compatible but most softphones do not have G.729 support. Paid softphones typically have support for G.729... --Tim
On Thu, Aug 19, 2010 at 10:28 PM, Zhang Shukun <bitzsk at gmail.com> wrote:> ?== 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? >http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt Then post your debug log, so we can see what is going on. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belanger at polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com