Thats really strange .. if you have made canreinvite=no then it should not
even attampt native bridging and should transcode codecs ..something's fishy
here .. Also try to put canreinvite=no in testulaw exntension too .
On 16/11/06, Victor Toofic <[EMAIL PROTECTED]> wrote:
El jue, nov 16 de 2006 a las 18:28 +0530, Vicky comentaba:
> g729 is not a free codec . YOu have to buy it from digium at rateof $10
per
> channel license . If you are just using asterisk and havent bought g729
> license then asterisk will just do bridging of g729 and wont
edit/transcode
> stream .
>
> On 16/11/06, Victor Toofic <[EMAIL PROTECTED]> wrote:
> >
> >I have the following scenario:
> >
> > g729 gsm
> > UAS <-----------> * <-----------> UAC
> >
> >I am using sipp to generate the calls between the UAC and the UAS and
> >sending some rtp from the UAC, I want * to do transcoding but as I see
> >it is not. As long as I know 'Attempting native bridge' means only
> >passing-through the rtp ¿Am I wrong?
I get the same message even if I'm not using g729:
--Attempting native bridge of SIP/testgsm-081784b0 and
SIP/testulaw-0817da80
ulaw gsm
UAS <-----------> * <-----------> UAC
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