<div dir="ltr">Yes, I agree. Tcpdump is one of my favourite programs. I need to enable it and disable it from the dialplan though.<div><br></div><div><br></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Fri, Feb 17, 2017 at 5:18 PM, Tim Pozar <span dir="ltr"><<a href="mailto:pozar@lns.com" target="_blank">pozar@lns.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">You can tell it to just capture SIP traffic and not the RTP traffic.<br>
Nice write up of using TCPdump and wireshark can be found here:<br>
<br>
<a href="https://blog.flowroute.com/2014/04/10/how-to-capture-sip-packets/" rel="noreferrer" target="_blank">https://blog.flowroute.com/<wbr>2014/04/10/how-to-capture-sip-<wbr>packets/</a><br>
<br>
BTW, I have found this works really well in trying to debug RTP traffic<br>
as well. Wireshark just does the right thing in putting audio back<br>
together. Very helpful in tracking down in and out of band DTMF<br>
problems that we were having with various carriers.<br>
<br>
Tim<br>
<span class=""><br>
On 2/17/17 3:07 PM, Derek Andrew wrote:<br>
> The SIP trace will be adequate but this is on a remote system with<br>
> limited disk space.<br>
><br>
> I would love to turn on debugging while making the troublesome calls,<br>
> then turn it off afterward.<br>
><br>
> Tcpdump is great, but starting it and stopping it and keeping all that<br>
> data would still be an issue.<br>
><br>
> d<br>
><br>
> On Fri, Feb 17, 2017 at 4:56 PM, Tim Pozar <<a href="mailto:pozar@lns.com">pozar@lns.com</a><br>
</span><span class="">> <mailto:<a href="mailto:pozar@lns.com">pozar@lns.com</a>>> wrote:<br>
><br>
> Why not capture the packets with something like tcpdump and run it<br>
> through Wireshark?<br>
><br>
> Tim<br>
><br>
> On 2/17/17 2:43 PM, Derek Andrew wrote:<br>
> > I have some troublesome numbers that I would like to capture the SIP<br>
> > dialogue when I am calling them. When I am about to dial the<br>
> number, is<br>
> > there any way to turn on SIP debugging in the dial plan before I make<br>
> > the call? (and turn it off after the call is completed?)<br>
> ><br>
> ><br>
> ><br>
> ><br>
> ><br>
><br>
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</div></div></blockquote></div><br><br clear="all"><div><br></div>-- <br><div class="gmail_signature" data-smartmail="gmail_signature"><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr">Copyright 2017 Derek Andrew (excluding quotations)<br><br>+1 306 966 4808</div><div dir="ltr"><font size="1">Communication and Network Services</font></div><div dir="ltr"><font size="1">Information and Communications Technology</font></div><div dir="ltr"><font size="1">Infrastructure Services<br></font><b>University of Saskatchewan<br></b><font size="1">Peterson 120; 54 Innovation Boulevard<br>Saskatoon,Saskatchewan,Canada. S7N 2V3<br>Timezone GMT-6<br></font><br>Typed but not read.<br><br><img src="http://homepage.usask.ca/dfa878/uofs.gif" width="200" height="15"><br><br></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div>
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