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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>How many g729 Licenses do you have? <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>Matteo Campana<br><b>Sent:</b> Wednesday, June 04, 2014 10:48 AM<br><b>To:</b> asterisk-users<br><b>Subject:</b> [asterisk-users] Renegotiate SIP audio codec after call is up<o:p></o:p></span></p></div><p class=MsoNormal><o:p> </o:p></p><div><p style='margin:0in;margin-bottom:.0001pt'><span style='font-size:10.5pt;font-family:"Lucida Grande","serif";color:#333333'>Hi All,<o:p></o:p></span></p><p style='margin:0in;margin-bottom:.0001pt'><span style='font-size:10.5pt;font-family:"Lucida Grande","serif";color:#333333'>Asterisk from 11.X branch is able to renegotiate an audio codec after a </span><span style='font-size:10.5pt;font-family:"Lucida Grande","serif";color:black'>SIP call session has been established (INVITE and 200 OK)?</span><span style='font-size:10.5pt;font-family:"Lucida Grande","serif";color:#333333'><o:p></o:p></span></p><p style='margin:0in;margin-bottom:.0001pt'><span style='font-size:10.5pt;font-family:"Lucida Grande","serif";color:#333333'><o:p> </o:p></span></p><p style='margin:0in;margin-bottom:.0001pt'><span style='font-size:10.5pt;font-family:"Lucida Grande","serif";color:black'>I have a problem with a reinvite sent by our SIP provider to change audio codec due to the recognition of a fax tone: after the call is established in g729, after a while I have the reinvite sent by the SIP provider with g711 in the SDP; Asterisk (v 1.4.33.1) says "Oooh, we need to change our audio formats since our peer supports only g729" and sends back 200 OK to the provider; at this point I have one no audio.</span><span style='font-size:10.5pt;font-family:"Lucida Grande","serif";color:#333333'><o:p></o:p></span></p><p style='margin:0in;margin-bottom:.0001pt'><span style='font-size:10.5pt;font-family:"Lucida Grande","serif";color:black'> </span><span style='font-size:10.5pt;font-family:"Lucida Grande","serif";color:#333333'><o:p></o:p></span></p><p style='margin:0in;margin-bottom:.0001pt'><span style='font-size:10.5pt;font-family:"Lucida Grande","serif";color:black'>So it seems that Asterisk responds 200 OK to the reinvite but really can not change the codec.</span><span style='font-size:10.5pt;font-family:"Lucida Grande","serif";color:#333333'><o:p></o:p></span></p><p style='margin:0in;margin-bottom:.0001pt'><span style='font-size:10.5pt;font-family:"Lucida Grande","serif";color:black'>Is that correct?</span><span style='font-size:10.5pt;font-family:"Lucida Grande","serif";color:#333333'><o:p></o:p></span></p><p style='margin:0in;margin-bottom:.0001pt'><span style='font-size:10.5pt;font-family:"Lucida Grande","serif";color:#333333'><o:p> </o:p></span></p><p style='margin:0in;margin-bottom:.0001pt'><span style='font-size:10.5pt;font-family:"Lucida Grande","serif";color:#333333'>Best regards,<o:p></o:p></span></p><p style='margin:0in;margin-bottom:.0001pt'><span style='font-size:10.5pt;font-family:"Lucida Grande","serif";color:#333333'>Matteo<o:p></o:p></span></p></div></div></body></html>