<div dir="ltr">Hi Chris, <div><br></div><div>I'm not using JSSIP or Sipml5, I'm using our own solution. As I said in the previous email, we are sending IP 0.0.0.0, port 0 and media inactive. It seems FS doesn't like this and I would like to know the reason.<div><br></div><div>I've changed our end to send<b> a=sendonly</b> (instead of<b> a=inactive</b>) and to not change the port to 0, and FS accepted it. The problem now is that after one second hearing the music on hold it stops, and I can never get the media back (I'm using TURN server to avoid NAT/Firewall problems).</div></div><div><br></div><div>Thank you.</div></div><div class="gmail_extra"><br><div class="gmail_quote">2015-07-23 18:11 GMT+02:00 Chris Tunbridge <span dir="ltr"><<a href="mailto:blasterjr@gmail.com" target="_blank">blasterjr@gmail.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Sergio, please fill in some additional information like which JS WebSocket client you're using, there are lots of hold issues with SipML5 and JsSIP, the only one that i know that has working hold music is sip.js (was a fork of JsSIP) and I've personally contributed towards making the hold function correctly on sip.js<br></div><div class="gmail_extra"><br><div class="gmail_quote"><div><div class="h5">On Thu, Jul 23, 2015 at 5:17 AM, Sergio García <span dir="ltr"><<a href="mailto:sergio.garcia@quobis.com" target="_blank">sergio.garcia@quobis.com</a>></span> wrote:<br></div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div class="h5"><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><span><font color="#888888">-- <br><div><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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</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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