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 <po...@lns.com
<mailto:po...@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.







--
_____________________________________________________________________
-- 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

Reply via email to