<div dir="ltr"><div>It wouldn&#39;t be the first time that a computer decided to behave because it knew Daddy was watching...<br></div>-MC<br></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Thu, May 16, 2013 at 7:20 PM, Sean Devoy <span dir="ltr">&lt;<a href="mailto:sdevoy@bizfocused.com" target="_blank">sdevoy@bizfocused.com</a>&gt;</span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div link="blue" vlink="purple" lang="EN-US"><div><p class="MsoNormal"><span style="font-family:&quot;Verdana&quot;,&quot;sans-serif&quot;;color:#1f497d">Thanks MC.  Had to load the pcapdev-lib, but got pcapsipdump installed.<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-family:&quot;Verdana&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-family:&quot;Verdana&quot;,&quot;sans-serif&quot;;color:#1f497d">My wife had just called my cell and got one way audio.  So I ran: pcapsipdump -f -p -i eth0 -d /tmp -n &lt;my number&gt;<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-family:&quot;Verdana&quot;,&quot;sans-serif&quot;;color:#1f497d">Of course I got 2 way audio.  I called the one that ALWAYS fails …. Got 2 way audio!  Does pcapsipdump fix it?  lol<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-family:&quot;Verdana&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-family:&quot;Verdana&quot;,&quot;sans-serif&quot;;color:#1f497d">I will try in day time tomorrow and see if we can get a failure.<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-family:&quot;Verdana&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-family:&quot;Verdana&quot;,&quot;sans-serif&quot;;color:#1f497d">Nothing in the freeswitch.log of value?  I didn’t see anything, but there is still a lot for me to learn there.<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-family:&quot;Verdana&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-family:&quot;Verdana&quot;,&quot;sans-serif&quot;;color:#1f497d">Sean<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-family:&quot;Verdana&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> <a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a> [mailto:<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a>] <b>On Behalf Of </b>Michael Collins<br>
<b>Sent:</b> Thursday, May 16, 2013 7:35 PM<br><b>To:</b> FreeSWITCH Users Help<br><b>Subject:</b> Re: [Freeswitch-users] One Way Audio<u></u><u></u></span></p><p class="MsoNormal"><u></u> <u></u></p><div><div><div><div><div>
<div><p class="MsoNormal" style="margin-bottom:12.0pt">Sean,<u></u><u></u></p></div><p class="MsoNormal" style="margin-bottom:12.0pt">Glad to hear you&#39;re making progress with using tcpdump and other packet capture-ish tools. You&#39;ve successfully captured the SIP call leg between your phone and your FreeSWITCH. That&#39;s good, but it&#39;s incomplete. You really want SIP and RTP both, and you want the call leg between FreeSWITCH and the telco. You have a few options:<u></u><u></u></p>
</div><p class="MsoNormal" style="margin-bottom:12.0pt">Expand your tcpdump. In other words, cast a wider net. Pro: easy to do. Con: creates massive pcap files through which you must sift to find the call in question.<u></u><u></u></p>
</div><p class="MsoNormal" style="margin-bottom:12.0pt">Use pcapsipdump. Pro: does all the work for you by putting SIP and RTP for each call leg into a single file. Cons: You have to compile it yourself, and it creates a whole lot of files, so you&#39;ll need to get used to it.<u></u><u></u></p>
</div><p class="MsoNormal" style="margin-bottom:12.0pt">My personal opinion is this: if you never, ever have to debug a SIP call ever again then just use tcpdump. However, if you&#39;re the phone guy and you&#39;ll be doing this again in the future then bite the bullet and learn pcapsipdump. Believe me it&#39;s worth it.<u></u><u></u></p>
</div><p class="MsoNormal">-MC<u></u><u></u></p></div><div><p class="MsoNormal" style="margin-bottom:12.0pt"><u></u> <u></u></p><div><p class="MsoNormal">On Thu, May 16, 2013 at 3:31 PM, Sean Devoy &lt;<a href="mailto:sdevoy@bizfocused.com" target="_blank">sdevoy@bizfocused.com</a>&gt; wrote:<u></u><u></u></p>
<div><p class="MsoNormal"><img src="cid:image001.gif@01CE5283.81116F80" border="0" height="31" width="31"><u></u><u></u></p><div><p class="MsoNormal"><span style="color:#1f497d">Hi all,</span><u></u><u></u></p><p class="MsoNormal">
<span style="color:#1f497d"> </span><u></u><u></u></p><p class="MsoNormal"><span style="color:#1f497d">First, I am on version: FreeSWITCH Version 1.2.5.3+git~20121219T035317Z~2b4aa48049 (git 2b4aa48 2012-12-19 03:53:17Z)  I hope to move to the Stable 1.2.9 this weekend.</span><u></u><u></u></p>
<p class="MsoNormal"><span style="color:#1f497d"> </span><u></u><u></u></p><p class="MsoNormal"><span style="color:#1f497d">I am having very reliable one way audio when calling Sprint Cell Phone users, though not every time.  I got this reproducible on my phone, but by the time I learned tcpdump command, it was working again.  However, the user that reported it seems to get it almost everytime.</span><u></u><u></u></p>
<p class="MsoNormal"><span style="color:#1f497d"> </span><u></u><u></u></p><p class="MsoNormal"><span style="color:#1f497d">Helpful tidbits:</span><u></u><u></u></p><p style="margin-left:40.5pt"><span style="font-family:Symbol;color:#1f497d">·</span><span style="font-size:7.0pt;color:#1f497d">         </span><span style="color:#1f497d">I THINK it happens in either direction.</span><u></u><u></u></p>
<p style="margin-left:40.5pt"><span style="font-family:Symbol;color:#1f497d">·</span><span style="font-size:7.0pt;color:#1f497d">         </span><span style="color:#1f497d">For this person at his home, it appears to be every time (for now)</span><u></u><u></u></p>
<p style="margin-left:40.5pt"><span style="font-family:Symbol;color:#1f497d">·</span><span style="font-size:7.0pt;color:#1f497d">         </span><span style="color:#1f497d">He reports calling (to or from) other Sprint Cell users results in the same problem from our FS</span><u></u><u></u></p>
<p style="margin-left:40.5pt"><span style="font-family:Symbol;color:#1f497d">·</span><span style="font-size:7.0pt;color:#1f497d">         </span><span style="color:#1f497d">It appears to only be true with Sprint Cell calls! (But my users say that’s not Sprints fault!)</span><u></u><u></u></p>
<p class="MsoNormal"><span style="color:#1f497d"> </span><u></u><u></u></p><p class="MsoNormal"><span style="color:#1f497d">Scenario:</span><u></u><u></u></p><p class="MsoNormal"><span style="color:#1f497d">I place a call from my Desk Cisco Phone (220) to his number 410493nnnn and it rings, he answers, I can hear him crystal clear … he can’t hear me at all.  I had a theory that it worked after 30 seconds (on my cell), but that does not hold true on his cell.</span><u></u><u></u></p>
<p class="MsoNormal"><span style="color:#1f497d"> </span><u></u><u></u></p><p class="MsoNormal"><span style="color:#1f497d">Here is the FS logfile:  </span><a href="http://www.bizfocused.com/Sean/fs_problem/freeswitch_no_audio_in.log.txt" target="_blank">http://www.bizfocused.com/Sean/fs_problem/freeswitch_no_audio_in.log.txt</a><u></u><u></u></p>
<p class="MsoNormal">And here is the tcpdump output:  <a href="http://www.bizfocused.com/Sean/fs_problem/dump.pcap.zip" target="_blank">http://www.bizfocused.com/Sean/fs_problem/dump.pcap.zip</a> <u></u><u></u></p><p class="MsoNormal">
 <u></u><u></u></p><p class="MsoNormal">Based on the small size of the file, I suspect someone is going to say “do it again with this tcpdump command”.  I welcome the education.<u></u><u></u></p><p class="MsoNormal"> <u></u><u></u></p>
<p class="MsoNormal">Anyway, any insight will be appreciated.<u></u><u></u></p><p class="MsoNormal"><span style="color:#1f497d"> </span><span style="color:#888888"><u></u><u></u></span></p><p class="MsoNormal"><span style="color:#1f497d">Sean</span><span style="color:#888888"><u></u><u></u></span></p>
</div></div><p class="MsoNormal" style="margin-bottom:12.0pt"><br>_________________________________________________________________________<br>Professional FreeSWITCH Consulting Services:<br><a href="mailto:consulting@freeswitch.org" target="_blank">consulting@freeswitch.org</a><br>
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</div><span class="HOEnZb"><font color="#888888"><p class="MsoNormal" style="margin-bottom:12.0pt"><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><a href="http://www.OSTAG.org" target="_blank">http://www.OSTAG.org</a><u></u><u></u></p></font></span></div></div></div><br>_________________________________________________________________________<br>

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<br></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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</div>