Hello
everyone,
Can someone tell me which annex the G.729 codec
from digium is.
Asterisk seems to thing it's Annex B (with a
warning in trasnlate.c)
[codec_g729b.so] => (Annex B (floating
point) G.729/PCM16 Codec Translator)
== Detected 10 licensed G.729 transcoders WARNING[8192]: 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 20 But the channels like IAX only work when you put in allow=G729 (without the B) When having the G729 code in the h323.conf and it's
building a connection with the H.323 channel i get:
2:51.058
ThreadID=0x00020011
h323caps.cxx(1626) H323 Added capability: G.729A{n/a}
<1>
2:51.059 ThreadID=0x00020011 h323caps.cxx(1687) H323 Found capability: G.729A{n/a} <1> I think this may be the source of the problems we
have with incomming H.323 call Audio only working one way...
(outgoing calls do fine though)
Is there just some inconsistency which needs
to be fixed, or is the codec an all G.729 codec which can do both A & B
? Or do i just have my H.323 allow=G729 wrong ?
Thanks in advance,
Tjardick van der Kraan
|
- [Asterisk-Users] Voicemail with H.323? Larry Creech
- Re: [Asterisk-Users] Licensed G.729 (from digi... Tjardick van der Kraan
- Re: [Asterisk-Users] Licensed G.729 (from ... Mark Spencer
- Re: [Asterisk-Users] Voicemail with H.323? Michael Manousos