[Asterisk-Users] G.729 Codec on Dialup

[Surajee Ratnayake]

----- Original Message -----
From: "Mark Spencer" <markster at digium.com>
To: <asterisk-users at lists.digium.com>
Sent: Wednesday, May 14, 2003 7:02 PM
Subject: Re: [Asterisk-Users] G.729 Codec on Dialup


> > On the local LAN, when we use the SJPhone as the SIP client,
communication works fine with no disturbances and noices. But when it comes
to dialup connection we harldy hear anything except a rough noice.
> > We have included G.729 Codec (Annex B) with the Asterisk server, and we
added the G.729 Codec to the SJPhone too. But it seems like SJPhone is using
a version of G.729 other than Codec G.729 (Annex B) , probably 'Annex A'.
> > What we want is to have a smoother communication when we use a SIP phone
on the Dialup connection.
>
> If you purchased G729 from Digium, you got the "Annex A" even though the
> file is, confusingly, called codec_g729b.so
>
> Mark
>

YES, we got the G729 from Digium.
but, when the asterisk is starting, its giving the following information,

"
 [codec_g729b.so] => (Annex B (floating point) G.729/PCM16 Codec Translator)
  == Detected 2 licensed G.729 transcoders
WARNING[1024]: File translate.c, Line 218 (calc_cost): Translator
'g729tolinb' does not produce sample frames.
  == Registered translator 'g729tolinb' from format 8 to 6, cost 99999
  == Registered translator 'lintog729b' from format 6 to 8, cost 46
"

so, isn't it using "Annex B"? if it is using "Annex B", how to get the
"Annex A"?

Thanx,
Surajee