Good day, I am trying to make tests and quality in Asterisk, is it possible to configure the Asterisk server to make about 100 simultaneous calls, without needing to configure softphone?
I have tried with .call files, but in the tests that I have done only works if the subscriber is configured in a softphone.
Channel: SIP/7001 Application: Playback Data: hello-world
 type: friend host: dynamic secrect=123 context=internal
exten => 7001,1,Answer exten => 7001,n,SetMusicOnHold(default) exten => 7001,n,WaitMusicOnHold(10) exten => 7001,n,Hangup
Asterisk has no knowledge whether a directly reachable peer is a hard phone or a soft phone, and the only way it knows the peer may be an intermediary, such as another PABX, or an ITSP, is that you don’t normally include any dialled digits in the dial string for peers that have direct user interfaces.
What do you mean by Asterisk server. Asterisk is a daemon, but you also need other software, on the same machine.
Any suggestions for quality testing?
For example, is it a good idea to use the SIPP program?
SIPP is very good software, you have to create XML configurations.