On Mon, Jun 20, 2011 at 11:58 PM, Matteo Campana <matteo.camp...@gmail.com>wrote:
> > > Inviato da iPhone > > Il giorno 18/giu/2011, alle ore 06:40, Larry Moore <lmo...@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<ewiel...@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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