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 <pozar at lns.com> 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/ > > 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. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20170217/186babc5/attachment.html>
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 <pozar at lns.com > <mailto:pozar at lns.com>> 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. > > > > >
Hi Derek, I think Homer (http://sipcapture.org/) is the right answer :-) HEP Agent will send the SIP trace to a remote Server (res_hep). Markus Am 18.02.2017 um 00:18 schrieb Tim Pozar:> 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 <pozar at lns.com >> <mailto:pozar at lns.com>> 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. >> >> >> >> >>