Yes, I agree. Tcpdump is one of my favourite programs. I need to enable it and disable it from the dialplan though.
On Fri, Feb 17, 2017 at 5:18 PM, Tim Pozar <[email protected]> wrote: > You can tell it to just capture SIP traffic and not the RTP traffic. > Nice write up of using TCPdump and wireshark can be found here: > > https://blog.flowroute.com/2014/04/10/how-to-capture-sip-packets/ > > BTW, I have found this works really well in trying to debug RTP traffic > as well. Wireshark just does the right thing in putting audio back > together. Very helpful in tracking down in and out of band DTMF > problems that we were having with various carriers. > > Tim > > On 2/17/17 3:07 PM, Derek Andrew wrote: > > The SIP trace will be adequate but this is on a remote system with > > limited disk space. > > > > I would love to turn on debugging while making the troublesome calls, > > then turn it off afterward. > > > > Tcpdump is great, but starting it and stopping it and keeping all that > > data would still be an issue. > > > > d > > > > On Fri, Feb 17, 2017 at 4:56 PM, Tim Pozar <[email protected] > > <mailto:[email protected]>> wrote: > > > > Why not capture the packets with something like tcpdump and run it > > through Wireshark? > > > > Tim > > > > On 2/17/17 2:43 PM, Derek Andrew wrote: > > > I have some troublesome numbers that I would like to capture the > SIP > > > dialogue when I am calling them. When I am about to dial the > > number, is > > > there any way to turn on SIP debugging in the dial plan before I > make > > > the call? (and turn it off after the call is completed?) > > > > > > > > > > > > > > > > > > > -- > > ____________________________________________________________ > _________ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com > -- > > > > Check out the new Asterisk community forum at: > > https://community.asterisk.org/ <https://community.asterisk.org/> > > > > New to Asterisk? Start here: > > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > <https://wiki.asterisk.org/wiki/display/AST/Getting+Started> > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > <http://lists.digium.com/mailman/listinfo/asterisk-users> > > > > > > > > > > -- > > Copyright 2017 Derek Andrew (excluding quotations) > > > > +1 306 966 4808 > > Communication and Network Services > > Information and Communications Technology > > Infrastructure Services > > *University of Saskatchewan > > *Peterson 120; 54 Innovation Boulevard > > Saskatoon,Saskatchewan,Canada. S7N 2V3 > > Timezone GMT-6 > > > > Typed but not read. > > > > > > > > > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk. > org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Copyright 2017 Derek Andrew (excluding quotations) +1 306 966 4808 Communication and Network Services Information and Communications Technology Infrastructure Services *University of Saskatchewan*Peterson 120; 54 Innovation Boulevard Saskatoon,Saskatchewan,Canada. S7N 2V3 Timezone GMT-6 Typed but not read.
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