Perfect, exactly what I needed. Thanks. On Mon, Feb 27, 2017 at 6:32 AM, Igor Zamocky <i...@zamocky.sk> wrote:
> Hi, > > If you are ok with starting debug via external system call, why not to use > something like this (I used to use something similar, it worked): > > exten =>* _XXX*,1,System(/usr/sbin/asterisk -rx ‘sip set debug peer *PEER* > ’) > same => n,Set(debug_on=1) > same => n,Dial(SIP/*PEER*/${EXTEN}) > > exten => *h*,1,GotoIf($[${debug_on} == 1]?undebug) > same => n,Hangup > same => n(undebug),System(( sleep 3 \; /usr/sbin/asterisk -rx 'sip set > debug off' ) &) > same => n,Set(debug_on=0) > same => n,Hangup > > I don’t know your setup, your dialplan logic, but I’m sure you can adapt > it to your needs. > > I. > > On 18 Feb 2017, at 00:26, Rafael dos Santos Saraiva <rafaels...@gmail.com> > wrote: > > Hi > > I don't know if works, but you can try this: > > System(tcpdump -nq -s 0 -i eth0 -w /tmp/sip.pcap port 5060 > or udp portrange 10000-20000 &); > Wait(1); > Dial(SIP/${EXTEN}); > System(pkill tcpdump); > Hangup; > > Or whitout RTP: > > System(tcpdump -nq -s 0 -i eth0 -w /tmp/sip.pcap port 5060 > &); > Wait(1); > Dial(SIP/${EXTEN}); > System(pkill tcpdump); > Hangup; > > Probably the last messages of SIP will be lost, BYE for example. > > > > > > 2017-02-17 20:43 GMT-02:00 Derek Andrew <derek.and...@usask.ca>: > >> 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 >> > > > > -- > Att, > Rafael Saraiva > -- > _____________________________________________________________________ > -- 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.
-- _____________________________________________________________________ -- 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