Hi, I just installed SIPP 3.3
Do you have a peer and extension configured for SIPP in your Asterisk configuration? You also needat least the -s option on your sipp command line. http://hasnainali.wordpress.com/2009/03/12/using-sipp-for-stress-testing-asterisk/ has some simple instructions which should get you started. If the calls still fail, Asterisk console output would be helpful.
I have a question here.
How can we test the quality of voice upon increasing the call load?
Can we try passing a voice file using sipp and record the same in dial plan record application ? Is this reliable enough to simulate near real world scenario?
Easily, as long as you have no media 🙂
Use -sn uac_pcap instead of -sn uac to test with RTP (and watch your call count drop). Add recording (MixMonitor()) to your dialplan and watch the call count go down even more. 😉
A rough way to see if call quality is deteriorating would be to call your Asterisk box while the SIPP test is running and listen to some message played via Background().