<div dir="ltr">Thanks Anthony.<div><br></div><div>I compiled OpenSSL v1.0.0l and re-compiled FreeSWITCH (--with-openssl) but I am still getting "Audio Handshake Failure 1" in switch_rtp.c as soon as a call is answered. Am I missing something? Would switching to Ubuntu 13.10 solve this issue?</div>
<div><br></div><div>Regards,</div><div>Kshitij</div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Fri, Feb 28, 2014 at 10:42 PM, Anthony Minessale <span dir="ltr"><<a href="mailto:anthony.minessale@gmail.com" target="_blank">anthony.minessale@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Its related to dtls make sure you are on the latest version which requires openssl not supported by that older ubuntu so you need to build your own. Use wireshark captures filtered on dtls to see the root of your problem.<div>
<br><div><br></div></div></div><div class="gmail_extra"><br><br><div class="gmail_quote"><div><div class="h5">On Fri, Feb 28, 2014 at 1:06 AM, Kshitij Saxena <span dir="ltr"><<a href="mailto:ksaxmail@gmail.com" target="_blank">ksaxmail@gmail.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"><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>
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<br></blockquote></div><span class="HOEnZb"><font color="#888888"><br><br clear="all"><div><br></div>-- <br><div dir="ltr">Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬<div><br><div>☞ <a href="http://freeswitch.org/" target="_blank">http://freeswitch.org/</a> ☞ <a href="http://cluecon.com/" target="_blank">http://cluecon.com/</a> ☞ <a href="http://twitter.com/FreeSWITCH" target="_blank">http://twitter.com/FreeSWITCH</a></div>
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