<div dir="ltr">Hello all,<div><br></div><div>I am using <b>FreeSwitch</b> as a <b>WebRTC gateway</b> thanks to its Websocket support (in my case WSS), but I'm facing this strange problem. Audio and Video calls are working perfectly fine, but when I try to set a call On Hold, FreeSwitch replies with "488 Not Acceptable Here" error to the ReINVITE I'm sending.</div><div><br></div><div>The only error I can see in the logs is:</div><div><br></div><div><font face="monospace, monospace"><b> [ERR] sofia.c:7280 Reinvite Codec Error!<br clear="all"></b></font><div><br></div><div>The only difference between the original INVITE and this ReINVITE is that I try to set IP address to <b>0.0.0.0</b>, port to <b>0</b> and media attribute to <b>inactive</b>. I don't understand what part of the SDP, FS doesn't "like".</div><div><br></div><div>Attached you can find a more detailed log file.</div><div><br></div><div>Thank you very much in advance.</div><div><br></div><div>Regards,</div>-- <br><div class="gmail_signature"><div dir="ltr"><div><div dir="ltr"><div><div><font color="#3d85c6"><b>Sergio García Ramos<br>
</b></font></div><div><font color="#666666">VoIP Engineer @ <a href="http://www.quobis.com/" target="_blank">Quobis</a> | e: <a href="mailto:sergio.garcia@quobis.com" target="_blank">sergio.garcia@quobis.com</a> </font><span style="color:rgb(102,102,102)"></span><span style="color:rgb(102,102,102)"> </span><span style="color:rgb(102,102,102)">| t:</span><span style="color:rgb(102,102,102)"> </span><a value="+34902999465">+34902999465</a><span style="color:rgb(102,102,102)"> </span><br><div><span><font color="#888888"><div style="font-size:11px;line-height:20px"><div><br></div>
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