<br><br><div class="gmail_quote">On Fri, Jul 22, 2011 at 4:46 PM, Eric Wieling <span dir="ltr"><<a href="mailto:EWieling@nyigc.com">EWieling@nyigc.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div lang="EN-US" link="blue" vlink="purple"><div><p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D"><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D">Asterisk does not support changing codecs on the fly.</span></p>
</div></div></blockquote><div><br></div><div><br></div><div>And why asterisk sends 200 OK to the provider, if does not support its re-invite?</div><div><br></div><div>M.</div><div><br></div><div><br></div><div> </div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div lang="EN-US" link="blue" vlink="purple"><div><p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D"><u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D"><u></u> <u></u></span></p>
<p class="MsoNormal"><b><span style="font-size:10.0pt">From:</span></b><span style="font-size:10.0pt"> <a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>] <b>On Behalf Of </b>Matteo Campana<br>
<b>Sent:</b> Friday, July 22, 2011 10:39 AM<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> [asterisk-users] Question about codec re-negotiation in asterisk 1.4.X<u></u><u></u></span></p>
<div class="im"><p class="MsoNormal"><u></u> <u></u></p><p class="MsoNormal">Hi all,<u></u><u></u></p><div><p class="MsoNormal">I have a major issue with a codec renegotiation in an asterisk 1.4.33.1 setup, which leads me to ask a general question about asterisk 1.4.X codec negotiation: asterisk can support a re-negotiation of a codec "on the fly" through a re-Invite? If my SIP provider sends me a re-invite changing codec from g729 to g711, asterisk properly handle the matter?<u></u><u></u></p>
</div><div><p class="MsoNormal">I see in the trace that asterisk responds 200 OK to the provider, but <u>does not send the re-invite to the UAC, and stops to send rtp to the UAC</u>.<u></u><u></u></p><p class="MsoNormal">
<span style="font-size:11.0pt;color:#1F497D"><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D"><u></u> <u></u></span></p></div></div></div></div><br>-</blockquote></div>