I would like to perform load test on asterisk with call recording feature.
I would be making outbound calls.
My system[PC] configuration:
OS : FC 6
Asterisk version: 1.2.13
Call Type: SIP to PSTN respondent
I would like to know how many sip peers can make simultaneous calls and receive audio clearly.
Please provide your suggestions for improving the performance on the same or how to conduct the same