[asterisk-users] No audio after a reinvite changing codec

[Matteo Campana]

On Mon, Jun 20, 2011 at 11:58 PM, Matteo Campana
<matteo.campana at gmail.com>wrote:

>
>
> Inviato da iPhone
>
> Il giorno 18/giu/2011, alle ore 06:40, Larry Moore <lmoore at starwon.com.au>
> ha scritto:
>
> > On 18/06/2011 5:36 AM, Matteo Campana wrote:
> >>
> >> Inviato da iPhone
> >>
> >> Il giorno 16/giu/2011, alle ore 16:37, Eric Wieling<EWieling at nyigc.com>
>  ha scritto:
> >>
> >>> We experience the same thing.  The solution we use is to not change
> codecs in the middle of a call.   I assumed it was an issue with our
> upstream.
> >>
> >> Hi Eric,
> >> this behavior  is an asterisk bug or asterisk can never change the codec
> "on the fly"?
> >>
> >>
> >> Thanks,
> >> Matteo
> >>
> >
> > The problem reported occurs after a fax tone is detected, one might
> expect T.38 or G711 to be used to handle the fax, not G729!
> >
> > There is no configuration file information for each of the nodes/peers,
> no debug of each peer involved  nor a trace of the packets hence no one will
> have ideas!
> >
> > Larry.
> >
>

 Hi,
I'm out of the office this week, next Monday I will send the debug to the
list.
However I think It's strange asterisk behavior: it says 200 OK after a
re-invite by the provider, but stops to send rtp.

Regards,

Matteo
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