Hello,
I am testing an ARI application written over Asterisk 13.6.0. In my ARI application, I wrote a bunch of features that manipulate call state, and I would now like to stress test it so that the application is production ready.
So, I am using SIPp, but I am having a LOT of trouble with docs.
Here's what I want to do, assuming each SIPp "peer" in the following description means a single SIPp process on, say, a linux host:
(a) REGISTER SIPp peer A and SIPp peer B to my Asterisk 13.6.0 using TCP transport using username and password.
(b) Use SIPp peer A to call SIPp peer B, both of whom are registered to my Asterisk 13.6.0/TCP PBX using INVITE.
(c) Wait for the call to be answered at peer B, and then pause 2 seconds and then kill the call by sending a BYE from SIPp peer B.
Here is what I am able to do so far: I am able to REGISTER the peers, but for the life of me, not able to get the peers to call each other.
I have tried sippy_cup, but I have had a similar issue where the documentation is spotty and I don't think it actually works for TCP. So that's not an option for me.
I would like to know how to create XML scenario files so that I can generate my own scenario files for my own set up from actual pjsip set logger on output.
Any help is deeply appreciated!
Thanks!
_______________________________________________ asterisk-app-dev mailing list asterisk-app-dev@xxxxxxxxxxxxxxxx http://lists.digium.com/cgi-bin/mailman/listinfo/asterisk-app-dev
