Hello, I am trying to run load on asterisk server(version 220.127.116.11) for the voicemail() application using SIPp tool. I am just running sipp at call rate of 1 cps with the following command: ./sipp -m 9000 -r 1 -rp 1000 -d 45 -max_socket 65536 -sf uac_msg_deposit.xml -i 172.16.129.13 -s 1 172.16.129.14 --trace_err I am trying to deposit 9000 messages in the mailbox of user 1 (given by the -s option) but the following warning is coming on the asterisk server due to which the message does not get deposited into the users mailbox: No audio available on SIP/172.16.129.13:5060-00000001?? I have…
Thank you for that information!
Has anyone else had an experience like this?
On 12 May 2011 20:25, Jared Geiger
> When I was testing 1.6.1 for high volume channels, I couldn't get over 1000
> channels / 40 CPS without the load average spiking up due to io wait. I
> switched back to 1.4 and I can go to 3000 channels / 75 CPS with no io wait
> and a load average in the 1s. It seemed like it was caused by…
Asterisk re-started automatically with 10 CPS(each call duration was 5sec)
Has someone met something similar ?
I could see in the log that asterisk segmentation fault and restart. Is there any idea how to regulate attack of invite to prevent restart? version 18.104.22.168
Astdatabase 5000 lines (using database 1 line every 1call)
cpu and memory usage is moderate(cpu MHz :1800.000 MemoryTotal:2075456 kB) thanks