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 On Thu, Jul 23, 2015 at 5:17 AM, Sergio García <sergio.garcia@quobis.com> wrote:
Hello all,
I am using *FreeSwitch* as a *WebRTC gateway* 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.
The only error I can see in the logs is:
* [ERR] sofia.c:7280 Reinvite Codec Error!*
The only difference between the original INVITE and this ReINVITE is that I try to set IP address to *0.0.0.0*, port to *0* and media attribute to *inactive*. I don't understand what part of the SDP, FS doesn't "like".
Attached you can find a more detailed log file.
Thank you very much in advance.
Regards, --
*Sergio García Ramos * VoIP Engineer @ Quobis <http://www.quobis.com/> | e: sergio.garcia@quobis.com | t: +34902999465
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