DHAVAL INDRODIYA
2009-Aug-31 06:36 UTC
[asterisk-users] SIPP how can we give delays between 2 calls
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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090831/38771097/attachment.htm
Alex Balashov
2009-Aug-31 07:08 UTC
[asterisk-users] SIPP how can we give delays between 2 calls
-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 > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-- 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
DHAVAL INDRODIYA
2009-Aug-31 08:45 UTC
[asterisk-users] SIPP how can we give delays between 2 calls
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 at 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 > > Register Now: http://www.astricon.net > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > 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 > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090831/69598ef3/attachment.htm