<div dir="ltr"><div><div>Hi Chris,<br><br>Welcome to the wacky world of telecom! It can be painful but when you finally get things working it's pretty awesome.<br><br>In your case I suspect a FreeSWITCH log of the call from start to finish would be most useful. A lot of info is available on this handy <a href="http://wiki.freeswitch.org/wiki/Reporting_Bugs#Debugging_Steps">wiki page</a>. I would bookmark that page if I were you because you'll probably use a lot of the tips presented there. Although the page is about "report bugs" it really starts with the debugging process itself which almost always means gathering information, analyzing it, and asking for help. <br>
<br></div>Using the information on that page, see if you can make a test call, get the FreeSWITCH log for that call, and put it on pastebin. The folks here will be happy to lend a hand with your analysis.<br><br></div><div>
<shameless_plug>You might also want to pick up the brand new <a href="http://www.packtpub.com/freeswitch-1-2/book">FreeSWITCH 1.2 book</a>.</shameless_plug><br><br></div>-MC<br><br></div><div class="gmail_extra">
<br><br><div class="gmail_quote">On Fri, May 24, 2013 at 8:14 AM, Chris Brannon <span dir="ltr"><<a href="mailto:chris@the-brannons.com" target="_blank">chris@the-brannons.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hello,<br>
I'm very new to freeswitch and telephony in general.<br>
I've managed to register phones with my server, but when I place test<br>
calls, the phone hangs, even though fs_cli tells me that the call was<br>
completed successfully.<br>
Also, everything looks fine when I run sip trace.<br>
FWIW, I've tried both linphone on Linux and csipsimple on Android.<br>
All of this is taking place on my LAN, so NAT isn't an issue.<br>
I started looking at traffic with tcpdump, and here's what I found.<br>
The phone authenticates to FreeSwitch and initiates the call.<br>
FreeSwitch sends a "100 trying" SIP message, followed by a UDP packet<br>
containing what appears to be SDP. It has no SIP header.<br>
This is followed by RTP from<br>
FreeSwitch to the phone. There is never a "200 OK" SIP message.<br>
<br>
Next, I ran linphone in debugging mode. I'll include a link to the log,<br>
in the hope that it will be helpful. Apparently, linphone isn't parsing<br>
the packet after the 100 trying message. Any idea what is going on<br>
here?<br>
<br>
<a href="http://the-brannons.com/linphone.log" target="_blank">http://the-brannons.com/linphone.log</a><br>
<br>
-- Chris<br>
<br>
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</blockquote></div><br><br clear="all"><br>-- <br>Michael S Collins<br>Twitter: @mercutioviz<br><a href="http://www.FreeSWITCH.org" target="_blank">http://www.FreeSWITCH.org</a><br><a href="http://www.ClueCon.com" target="_blank">http://www.ClueCon.com</a><br>
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