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

[Matteo Campana]

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