<div dir="ltr"><div><font color="#444444" face="Calibri" size="3"><span style="line-height:22.719999313354492px">Summary: SIPml / JSSIP calls over WS to FreeSWITCH are terminated with "Audio Handshake Failure 1" in switch_rtp.c</span></font></div>
<span style="color:rgb(68,68,68);font-family:Calibri;font-size:16px;line-height:22.719999313354492px"><div><span style="color:rgb(68,68,68);font-family:Calibri;font-size:16px;line-height:22.719999313354492px"><br></span></div>
I am running FreeSWITCH Version 1.5.8b on an Ubuntu 12.04 server with a Sangoma A101DE PRI card running on Wanpipe 7.0.10. The PRI card is connected to the ISP's network.</span><div style="line-height:22.719999313354492px;color:rgb(68,68,68);font-family:Calibri;font-size:16px">
<div><br></div><div>When I try connecting to FreeSWITCH from the SIPml or JSSIP demo pages, (over websocket, RTCWebBreaker enabled, port 5066 open on server for tcp/udp along with 5060-5081 and 16383-32768), the client is successfully registered, but any attempt to make a call is failing. In fs_cli I can see an error "Audio Handshake Failure 1" in switch_rtp.c as soon as a call is attempted from the client.</div>
<div><br></div><div>When using X-lite, the calls are successful. Originate from within fs_cli is also successful.</div><div><br></div><div>I have been splitting my hair for 2 days but cannot begin to decipher what has gone wrong! What is worse is that I have used the exact same configuration successfully at 2 previous locations!</div>
<div><br></div><div>Any help would be very greatly appreciated.</div><div><br></div><div>Full Version: FreeSWITCH Version 1.5.8b+git~20140208T085053Z~4fa68fcd75~64bit (git 4fa68fc 2014-02-08 08:50:53Z 64bit)<br>Originate: originate freetdm/1/a/99xxxxxxxx &bridge(freetdm/1/a/99xxxxxxxx)</div>
<div><br></div><div>Regards,</div><div>Kshitij Saxena</div></div></div>