Hi all of you,
i want to generate tls calls using sipp with asterisk pbx,
can any one please guide, how to configure tls support with asterisk and
generate calls
Thanks
Kumar
On Fri, Apr 17, 2015 at 9:17 PM, Kai Panzinger <[email protected]> wrote:
> Hi, the xml can be found in my first post. I am using the g722 pcap that
> is provided with sipp. I tried around a bit and after removing the
> format-information from the xml it works fine.
>
> The command i am using is:
> Sipp -sf raduas.xml -i 192.168.40.1
>
> I tried it out on the client side with:
> Sipp -sn uac 192.168.40.1 -d 10000
>
> In my test case it worked just fine now. On monday i will most probably
> have a chance to try it out with one of our corporate operator positions on
> the client side. I will let you know if it worked well then.
>
> Thanks kai
> ------------------------------
> Von: Sakharam Thorat <[email protected]>
> Gesendet: 17.04.2015 17:36
> An: Kai Panzinger <[email protected]>
> Cc: [email protected]; sipp-users <[email protected]>
> Betreff: RE: [Sipp-users] No valid To: header in reply
>
> please attach .pcap and .xml file also post command you using to run sipp
>
>
> Best Regards,
> Sakharam Thorat.
>
>
> ------------------------------
> Date: Fri, 17 Apr 2015 09:56:52 +0200
> Subject: Re: [Sipp-users] No valid To: header in reply
> From: [email protected]
> To: [email protected]
> CC: [email protected]; [email protected]
>
> Sorry,
>
> here it is.
>
> 2015-04-17 7:24 GMT+02:00 Sakharam Thorat <[email protected]>:
>
>
> Kai , I think you you missed to attach trace
>
> Best Regards,
> Sakharam Thorat.
>
>
> ------------------------------
> Date: Thu, 16 Apr 2015 13:53:06 +0200
> From: [email protected]
> To: [email protected]
> CC: [email protected]
> Subject: Re: [Sipp-users] No valid To: header in reply
>
>
> Hi,
>
> here is the trace you asked for.
> As far as i can see now the problem results out of the fact that the
> format-information of the .xml is transmitted as well.
> I think you should also see it when you have a look into the content of
> the 200 OK. I will let you know if this solves my issue.
>
> Thanks for now,
> Kai
>
> 2015-04-15 16:24 GMT+02:00 <[email protected]>:
>
> Hi Kai,
>
> Could you please provide sip traces?
>
> Thanks, v.
>
>
> Sent by Outlook <http://taps.io/outlookmobile> for Android
> _____________________________
> From:Kai Panzinger
> Subject:[Sipp-users] No valid To: header in reply
> To:[email protected]
>
>
>
> Hey Guys,
>
> I am Using sipp to simulate a Radiostation in an IP-Radio Environment.
> As such a radio is compliant to the Eurocae standard it has to act like
> the following:
> ---recv---> INVITE //Basestation connects to the radiochannel
> <---send--- 200 OK
> ---recv---> ACK
> <--transmition-->
> ---recv---> BYE //Basestation disconnects from the
> radiochannel
> <---send--- 200 OK
>
> In order to do this i have written an .xml for simulating the (UAS)
> radiostation (Script is in appendance).
> For testing i am feeding my server with the standard UAC procedure. In
> final state i will connect my UAS to the corporate Radio Basestation
> Equipment my company developes.
>
> Problem is that the UAC throws the error (in subject) as soon as my UAS
> replies with its first 200.
> Hope you guys can help me with this matter.
>
>
>
>
>
>
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--
Kumar
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Develop your own process in accordance with the BPMN 2 standard
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