Welcome to the FreeSWITCH community!<div><br></div><div>Okay, in cases like this you will need to break out the information gathering tools. Start with the stuff here:</div><div><br></div><div><a href="http://wiki.freeswitch.org/wiki/Reporting_Bugs">http://wiki.freeswitch.org/wiki/Reporting_Bugs</a></div>
<div><br></div><div>You'll need a FreeSWITCH console log. You might also want to do a tcpdump on the RTP and see if it's really flowing. Hopefully that will yield a clue as to what you need to do...</div><div><br>
</div><div>You can also dump all this into pastebin and put the URL in this thread. The link I provided has all the information you need on how to collect information and put it on pastebin.</div><div><br></div><div>Hope it goes well!</div>
<div><br></div><div>-MC<br><br><div class="gmail_quote">On Fri, Apr 22, 2011 at 7:38 PM, Chris Monkey <span dir="ltr"><<a href="mailto:ckmonkey158@yahoo.com">ckmonkey158@yahoo.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<table cellspacing="0" cellpadding="0" border="0"><tbody><tr><td valign="top" style="font:inherit">Hi,<br><br>I am am relatively new to FS and have setup a home PBX with
about 5 Snom 320s with TLS. Everything works fine, except when I try to
use the mad_boss example, or any auto_outcall conference, the phones
ring and will auto answer, but there is no audio. However, if I dial in
to the conference, everything works fine.<br><br>I've tried adding
<action application="set" data="sip_secure_media=true"/> to the
conference extension, but to no avail. I don't see any errors in the
log, and I just did a make clean this afternoon. What am I missing?<br><br>Thanks</td></tr></tbody></table><br>_______________________________________________<br>
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