Re: [asterisk-users] Turn on SIP debugging from DialPlan

2017-02-27 Thread Derek Andrew 
Perfect, exactly what I needed. Thanks.

On Mon, Feb 27, 2017 at 6:32 AM, Igor Zamocky  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*/${EXTEN})
>
> exten => *h*,1,GotoIf($[${debug_on} == 1]?undebug)
> same => n,Hangup
> same => n(undebug),System(( sleep 3 \; /usr/sbin/asterisk -rx 'sip set
> debug off' ) &)
> same => n,Set(debug_on=0)
> same => n,Hangup
>
> I don’t know your setup, your dialplan logic, but I’m sure you can adapt
> it to your needs.
>
> I.
>
> On 18 Feb 2017, at 00:26, Rafael dos Santos Saraiva 
> wrote:
>
> Hi
>
> I don't know if works, but you can try this:
>
> System(tcpdump -nq -s 0 -i eth0 -w /tmp/sip.pcap port 5060
> or udp portrange 1-2 &);
> Wait(1);
> Dial(SIP/${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 :
>
>> 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

Re: [asterisk-users] Turn on SIP debugging from DialPlan

2017-02-27 Thread Igor Zamocky 
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  
> 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 1-2 &);
> 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  >:
> 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

-- 
_
-- 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

Re: [asterisk-users] Turn on SIP debugging from DialPlan

2017-02-18 Thread Markus 
While we're at it, check out sngrep. Alex B. mentioned it on another 
mailing list a couple days ago.


Screenshots: https://github.com/irontec/sngrep/wiki/Screenshots

Download: https://github.com/irontec/sngrep


Am 18.02.2017 um 05:10 schrieb Markus Weiler:

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 > 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.











--
_
-- 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

Re: [asterisk-users] Turn on SIP debugging from DialPlan

2017-02-17 Thread Markus Weiler 

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 > 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.








--
_
-- 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

Re: [asterisk-users] Turn on SIP debugging from DialPlan

2017-02-17 Thread Steve Edwards 

On Fri, 17 Feb 2017, 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?)


You could use the system() application as suggested before.

You could also just start a console packet logger and just leave it 
running:


sudo ngrep -O ngrep.pcap -W byline -d any  port 5060

This will only capture packets containing your ANI which includes INVITE, 
Trying, OK, ACK, and BYE -- basically, the entire SIP dialog for the call.


This will only take about 4kB per call, so you can log over 250 calls per 
mega-byte so I'm guessing that should be possible.


--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281

--
_
-- 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

Re: [asterisk-users] Turn on SIP debugging from DialPlan

2017-02-17 Thread Derek Andrew 
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  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  > > 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.
> >
> >
> >
> >
> >
>
> --
> _
> -- 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

Re: [asterisk-users] Turn on SIP debugging from DialPlan

2017-02-17 Thread Derek Andrew 
But how do you turn on the debugging from the dialplan? What would be cool
is:

same => n,TURN ON DEBUGGING



On Fri, Feb 17, 2017 at 5:09 PM, Victor Villarreal 
wrote:

> Hi Derek,
>
> SIP debug can be enabled via Asterisk CLI (console) with the command:
>
> asterisk> sip set debug on
>
> If you know via what trunk your call goes, you can use the following
> command instead:
>
> asterisk> sip set debug ip xxx.xxx.xxx.xxx
>
> Where the xxx is the IP of your trunk (voip to pstn provider).
>
> Affter you make all your test, simply issue:
>
> asterisk> sip set debug off
>
> And all the SIP conversation are saved in your full log file.
>
> More info here:
>
> https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
>
> If what you want is test your dialplan, simply use the command:
>
> asterisk> dialplan show xxx@your_context
>
> Where xxx is the number you want to dial, from the context asigned to your
> extension.
>
> Cheers
>
>
> El 17/2/2017 19:44, "Derek Andrew"  escribió:
>
>> 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.
-- 
_
-- 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

Re: [asterisk-users] Turn on SIP debugging from DialPlan

2017-02-17 Thread Rafael dos Santos Saraiva 
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 1-2 &);
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 :

> 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

Re: [asterisk-users] Turn on SIP debugging from DialPlan

2017-02-17 Thread 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  > 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.
> 
> 
> 
> 
> 

-- 
_
-- 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

Re: [asterisk-users] Turn on SIP debugging from DialPlan

2017-02-17 Thread Victor Villarreal 
Hi Derek,

SIP debug can be enabled via Asterisk CLI (console) with the command:

asterisk> sip set debug on

If you know via what trunk your call goes, you can use the following
command instead:

asterisk> sip set debug ip xxx.xxx.xxx.xxx

Where the xxx is the IP of your trunk (voip to pstn provider).

Affter you make all your test, simply issue:

asterisk> sip set debug off

And all the SIP conversation are saved in your full log file.

More info here:

https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

If what you want is test your dialplan, simply use the command:

asterisk> dialplan show xxx@your_context

Where xxx is the number you want to dial, from the context asigned to your
extension.

Cheers


El 17/2/2017 19:44, "Derek Andrew"  escribió:

> 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
>
-- 
_
-- 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

Re: [asterisk-users] Turn on SIP debugging from DialPlan

2017-02-17 Thread Derek Andrew 
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  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.
-- 
_
-- 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

Re: [asterisk-users] Turn on SIP debugging from DialPlan

2017-02-17 Thread Tim Pozar 
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

[asterisk-users] Turn on SIP debugging from DialPlan

2017-02-17 Thread Derek Andrew 
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