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...have given the rtp port range as 6000 to 8000 in rtp.conf. Is this not sufficient for running 1000 calls. The SIPp command which I am running ?is as follows: /sipp -m 10 -r 1 -rp 1000 -d 45 -mp 7000 -p 9000 -bg -max_socket 65536 -sf uac_msg_deposit.xml -i 172.16.129.13 -s 1 172.16.129.14 --trace_err usleep 80000 ./sipp -m 10 -r 1 -rp 1000 -d 45 -mp 7004 -p 9001 -bg -max_socket 65536 -sf uac_msg_deposit.xml -i 172.16.129.13 -s 2 172.16.129.14 --trace_err usleep 80000 ./sipp -m 10 -r 1 -rp 1000 -d 45 -mp 7008 -p 9002 -bg -max_socket 65536 -sf uac_msg_deposit.xml -i 172.16.129.13 -s 3 172.16....

...000????????????? | 00:00:31:509???????????? ------------------------------ Test Terminated -------------------------------- 2013-05-20?22:55:17:675?1369083317.675242: Aborting call on UDP retransmission timeout for Call-ID '120-60749 at 192.168.1.114'. sipp: There were more errors, enable -trace_err to log them. This an error message I get when I use -trace_err: 2013-05-20?23:00:59:021??? 1369083659.021771: Aborting call on UDP retransmission timeout for Call-ID '33-60833 at 192.168.1.114 Thanks in advance. Regards, Tom -------------- next part -------------- An HTML attachment was scr...

Hi friends... I am trying to test my voicemail with Asterisk using SIPP (SIPP is running in Debian Squezze and Asterisk is running in OpenSuSE-11.1), the command that I use is: sipp -sn uac_pcap -l 1 -m 1 -s 55 -trace_err 192.168.13.6 But, If I use the file g711a.pcap included in the sources of sipp or if use some file captured for me the result is the same ---> error ... the message in Asterisk is: [Apr 2 02:16:14] WARNING[21197]: app.c:674 __ast_play_and_record: No audio available on SIP/sipp-082402b0??...

Hello, I have asterisk 1.4.4 running with anonymous sip calls enabled and I am testing the box for load using sipp with something like this - sipp -sn uac -s 10 -d 60000 -i 192.168.1.49 -l 110 -r 5 -trace_err 192.168.1.50 Asterisk picks up the call and runs a test php-agi file that plays a .gsm file. As soon as the number of active calls reaches 99, asterisk starts Declining further calls. I thought this was a call-limit problem so created a type=friend entry for the sipp client's IP and it still...

We are in the process of implementing realtime voicemail. I was wanting to "stress-test" the system to see if or when it would fall over. Is it possible to use sipp to create say 250 calls, each of which leaves a message in the voicemail ? My dialplan is currently [default] exten => stress,1,Answer() exten => stress,2(vm),Voicemail(7777|su) exten => stress,3,Hangup()

Hi, I'd like to test Asterisk performance under more concurrent sip calls. I use Sipp, but do get "Unexpected message for Call-ID ...", so I wonder if anyone is using sipp succesfully with Asterisk and is willing to share more info about his solution ... Any other convenient way to load test Asterisk ? Is sipp the right tool ? Thanks in advance, regards, Rob. sipp: The

Hello Everyone, We are getting some rather poor results (relative) with our Asterisk setup. Not sure if we are using the sipp correctly etc.. but nevertheless, is there any documentation that describes how we can get the most our of our Asterisk box. For example when we hit the "too many file" error, and fixing it using ulimit..... Also, is there any way we can allocate sufficient

Hello everyone I am trying to look for a free test call generator that will get me some stats like PDD, ASR and call quality etc on each route. As well as do test at every interval too If you know something like this please enlighten me. Sam -------------- next part -------------- An HTML attachment was scrubbed... URL: