HI list,<div>no idea?? :)</div><div><br></div><div>M.<br><br><div class="gmail_quote">On Mon, Jun 13, 2011 at 6:55 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;">Hi all,<br>
we have a problem with a reinvite sent by our SIP provider to change audio codec due to the recognition of a fax tone.<br>
After that the SIP call session has been established (INVITE and 200 OK) we have the following codec situation: <br>
<br>
UAC ASTERISK UAS | ASTERISK UAC PROVIDER<br>
g711 <----------------------> g711 | g729 <---------------------------> g729 <br>
rtp rtp<br>
<br>
After a while, we have the reinvite sent by the SIP provider with g711 in the SDP.<br>
So asterisk need to change audio codec from g729 to g711 and correctly we see on debug the following line:<br>
"Oooh, we need to change our audio formats since our peer supports only g729" and asterisk send back 200 OK to the provider.<br>
At this point we have one way rtp audio:<br>
<br>
UAC ASTERISK UAS | ASTERISK UAC PROVIDER<br>
g711 ----------------------> g711 | g711 ---------------------------> g711 <br>
rtp rtp<br>
<br>
So the problem is that UAC does not hear audio at all.<br>
Any idea?<br>
<br>
(Asterisk version: 1.4.33.1).<br>
<br>
Thanks in advance,<br><font color="#888888">
Matteo
</font></blockquote></div><br></div>