<br><br><div class="gmail_quote">On Mon, Jun 20, 2011 at 11:58 PM, Matteo Campana <span dir="ltr"><<a href="mailto:matteo.campana@gmail.com">matteo.campana@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<br>
<br>
Inviato da iPhone<br>
<br>
Il giorno 18/giu/2011, alle ore 06:40, Larry Moore <<a href="mailto:lmoore@starwon.com.au">lmoore@starwon.com.au</a>> ha scritto:<br>
<div class="im"><br>
> On 18/06/2011 5:36 AM, Matteo Campana wrote:<br>
>><br>
>> Inviato da iPhone<br>
>><br>
>> Il giorno 16/giu/2011, alle ore 16:37, Eric Wieling<<a href="mailto:EWieling@nyigc.com">EWieling@nyigc.com</a>> ha scritto:<br>
>><br>
>>> 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.<br>
>><br>
>> Hi Eric,<br>
>> this behavior is an asterisk bug or asterisk can never change the codec "on the fly"?<br>
>><br>
>><br>
>> Thanks,<br>
>> Matteo<br>
>><br>
><br>
> 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!<br>
><br>
> 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!<br>
><br>
> Larry.<br>
><br></div></blockquote><div><br></div><div> Hi,</div><div>I'm out of the office this week, next Monday I will send the debug to the list.</div><div>However I think It's strange asterisk behavior: it says 200 OK after a re-invite by the provider, but stops to send rtp.</div>
<div> </div><div>Regards,</div><div><span class="Apple-style-span" style="color: rgb(136, 136, 136); "><br></span></div><div><span class="Apple-style-span" style="color: rgb(136, 136, 136); ">Matteo</span></div></div><br>