'Re: [asterisk-users] SIPP how can we give delays between 2 calls'

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List:       asterisk-users
Subject:    Re: [asterisk-users] SIPP how can we give delays between 2 calls
From:       DHAVAL INDRODIYA <dhaval.it01034 () gmail ! com>
Date:       2009-08-31 8:57:55
Message-ID: cf554c8c0908310145m26e407e2se502b33ff78d97d9 () mail ! gmail ! com
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thanks Alex,

it works

but can you tell me about any sound playing on SIPP means ,
once SIPP channels  connect in conference room then there is lots of noise ,

is there any way to reduce it.

regards
Dhaval

On Mon, Aug 31, 2009 at 12:38 PM, Alex Balashov
<abalashov@evaristesys.com>wrote:

> -r is a flag that regulates the call setup rate per second.
>
> DHAVAL INDRODIYA wrote:
>
> > hello,
> >
> > i am using following SIPP command to test My meetme conference
> >
> > ./sipp -sn uac -d 300000 -s 8600 127.0.0.1 -l 20
> >
> >
> > which generates 20 call to my server but i need to give delay between
> > each call
> >
> > once 1 st call is placed then second call should be placed after few
> > seconds
> >
> > and is there any method to play some file file or data while SIPP call
> > is placed
> >
> > i got very bad sound while sipp  calls connect to my meetme room
> >
> > can any body have idea regarding this ,
> >
> > regards
> > Dhaval
> >
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > AstriCon 2009 - October 13 - 15 Phoenix, Arizona
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>
>
> --
> Alex Balashov - Principal
> Evariste Systems
> Web    : http://www.evaristesys.com/
> Tel    : (+1) (678) 954-0670
> Direct : (+1) (678) 954-0671
> Mobile : (+1) (678) 237-1775
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>

[Attachment #5 (text/html)]

thanks Alex,<br><br>it works <br><br>but can you tell me about any sound playing on \
SIPP means ,<br>once SIPP channels  connect in conference room then there is lots of \
noise ,<br><br>is there any way to reduce it.<br><br> regards<br>Dhaval<br><br><div \
class="gmail_quote">On Mon, Aug 31, 2009 at 12:38 PM, Alex Balashov <span \
dir="ltr">&lt;<a href="mailto:abalashov@evaristesys.com">abalashov@evaristesys.com</a>&gt;</span> \
wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, \
                204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
-r is a flag that regulates the call setup rate per second.<br>
<div><div></div><div class="h5"><br>
DHAVAL INDRODIYA wrote:<br>
<br>
&gt; hello,<br>
&gt;<br>
&gt; i am using following SIPP command to test My meetme conference<br>
&gt;<br>
&gt; ./sipp -sn uac -d 300000 -s 8600 127.0.0.1 -l 20<br>
&gt;<br>
&gt;<br>
&gt; which generates 20 call to my server but i need to give delay between<br>
&gt; each call<br>
&gt;<br>
&gt; once 1 st call is placed then second call should be placed after few<br>
&gt; seconds<br>
&gt;<br>
&gt; and is there any method to play some file file or data while SIPP call<br>
&gt; is placed<br>
&gt;<br>
&gt; i got very bad sound while sipp  calls connect to my meetme room<br>
&gt;<br>
&gt; can any body have idea regarding this ,<br>
&gt;<br>
&gt; regards<br>
&gt; Dhaval<br>
&gt;<br>
&gt;<br>
</div></div>&gt; ------------------------------------------------------------------------<br>
 &gt;<br>
&gt; _______________________________________________<br>
&gt; -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" \
target="_blank">http://www.api-digital.com</a> --<br> &gt;<br>
&gt; AstriCon 2009 - October 13 - 15 Phoenix, Arizona<br>
&gt; Register Now: <a href="http://www.astricon.net" \
target="_blank">http://www.astricon.net</a><br> &gt;<br>
&gt; asterisk-users mailing list<br>
&gt; To UNSUBSCRIBE or update options visit:<br>
&gt;    <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" \
target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br> <br>
<br>
--<br>
Alex Balashov - Principal<br>
Evariste Systems<br>
Web    : <a href="http://www.evaristesys.com/" \
target="_blank">http://www.evaristesys.com/</a><br> Tel    : (+1) (678) 954-0670<br>
Direct : (+1) (678) 954-0671<br>
Mobile : (+1) (678) 237-1775<br>
<br>
_______________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" \
target="_blank">http://www.api-digital.com</a> --<br> <br>
AstriCon 2009 - October 13 - 15 Phoenix, Arizona<br>
Register Now: <a href="http://www.astricon.net" \
target="_blank">http://www.astricon.net</a><br> <br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
   <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" \
target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br> \
</blockquote></div><br>

_______________________________________________
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