[asterisk-users] how to know RTP por of a SIP client in

[Sammy Govind]

OP may be able to use System through Dial plan but I'm thinking that since
tcpdump don't just give output within seconds or neither do it get
daemonized? so this system() call will hold the call to that priority. This
may even result in call failure. I think this system call should trigger a
shell script that launches an instance of tcpdump and move forward in the
dial plan.

Can anyone tell if we can extract a header value from SDP(for RTP Tx/RX
ports) within the SIP packet using the SIP_HEADER function?

How about using sipgrep: The idea is launch a sipgrep based scripts in the
background which just takes Call-ID and parse RTP port data and save it in
memcached. This memchache Key/value register will just save [Call-ID:RTP
port data] for each call entering into the Server. This script should start
separate instances of tcpdump for each call with separate file names.

On each call hangup call the h extensions will use the SIP_HEADER(call-id)
Key and trigger a stop command for the background tcpdump for this
particular call.


On Mon, Oct 24, 2011 at 4:36 AM, Bruce B <bruceb444 at gmail.com> wrote:

> Then you may use system() in dial-plan to run that shell command along with
> what I suggested.
>
> -Bruce
>
>
> On Sun, Oct 23, 2011 at 5:22 AM, ISABEL ORDAS ARNAL <ioa at tid.es> wrote:
>
>>
>> Yes, I need to know to get in in dialplan because I want to capture
>> traffic per call. I would like to launch $SHELL{tcpdump src port xxxx} in
>> the dialplan or something like this. And I want RTP traffic only of a
>> certain call.
>> Thank you!
>>
>> =======================
>> Date: Fri, 21 Oct 2011 09:41:39 -0400
>> From: Bruce B <bruceb444 at gmail.com>
>> Subject: Re: [asterisk-users] how to know RTP por of a SIP client in
>>        the     dialplan
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>>        <asterisk-users at lists.digium.com>
>> Message-ID:
>>        <CAJyE_uWLRXkhrWQ-6SvNW1ihN-nGA3HFwHt=
>> PU-tfR6LYbi5mg at mail.gmail.com>
>> Content-Type: text/plain; charset="iso-8859-1"
>>
>> Do you need to know to get it in dialplan? If I not, from shell (not
>> Asterisk CLI) I usually use:
>>
>> netstata -a | grep asterisk
>>
>> By default Asterisk settings it should be something between 10k-20k
>>
>> -Bruce
>>
>> On Fri, Oct 21, 2011 at 3:46 AM, ISABEL ORDAS ARNAL <ioa at tid.es> wrote:
>>
>> >  Hi all, ****
>> >
>> > How can I get the RTP port one SIP client is using for sending/receiving
>> > RTP flow? Can I obtain it in from SIP_HEADER of something like that in
>> the
>> > dialplan?****
>> >
>> > Thank you!****
>> >
>> > ** **
>> >
>> > Isabel****
>> >
>>
>>
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